| 00:00.05 | sawgood | This might be the entire fix! |
| 00:00.38 | p3nguin | externip needs to be your external IP address, and if that external IP address is not on Asterisk, then you shouldn't be putting Asterisk's IP address in that setting. |
| 00:01.07 | sawgood | yes, Asterisk is setup and working as a NAT router |
| 00:01.21 | p3nguin | It is on the perimeter, now? |
| 00:01.32 | sawgood | 100% corret (it was not until today) |
| 00:01.39 | sawgood | I put in a 2nd NIC card with a static IP address |
| 00:01.47 | p3nguin | And you've got a static IP address from your ISP? |
| 00:02.04 | sawgood | NIC 1 = 192.168.1.x ... NIC 2 = static IP from ISP |
| 00:02.19 | sawgood | I configured Asterisk in CentOS to be a NAT router and masqurade |
| 00:02.20 | p3nguin | Sounds okay to me. |
| 00:02.36 | sawgood | This was not the case until today ... |
| 00:09.35 | sawgood | now to test the phones ... |
| 00:09.46 | sawgood | all has been rebooted with the right setting(s) |
| 00:09.52 | p3nguin | Can anyone explain to me why and how my voice mail notifications (in my email) reflect the wrong message number and have an attachment which does not exist? It says new message 12 and has msg0012.WAV attached. I check in my INBOX on the server and msg0011 is the last one, which also corresponds with the time on the email. |
| 00:11.38 | *** join/#asterisk Janos (~Janos@190.10.52.113) |
| 00:11.47 | p3nguin | I played the attachment in the email and it is the last email in my INBOX, which is msg0011.WAV. How is the email able to say it's message 12 and have msg0012.WAV if that's not really the message number nor file name? |
| 00:12.27 | Janos | hello, can someone give me a link to the documentation on how to use lua as dialplan programing language please |
| 00:20.31 | sawgood | Calls seem to not be dropping now ... |
| 00:20.39 | sawgood | Thanks for your advice and help! |
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| 00:54.14 | pabelanger | p3nguin: what version of Asterisk? |
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| 00:59.13 | p3nguin | 1.4.40 |
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| 01:43.19 | p3nguin | Is there a setting for the voicemail sound files' permissions? -rwxr-xr-x is not an appropriate mask for a sound file, so I need to change or set it if possible. |
| 01:48.40 | p3nguin | This doesn't make any sense. I just left myself a voicemail, and it is msg0012.WAV on the file system. I got the email notification, and it says message 13 and msg0013.WAV is attached. WTF is the problem? |
| 01:51.20 | pabelanger | p3nguin: I suspect this is a bug. alecdavis in #asterisk-dev reported something very similar. In his case, Asterisk was re-recording over voicemail messages. I'm wondering if you are seeing the same thing |
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| 01:58.34 | p3nguin | I've never noticed this before, but I also had voicemail email notifications disabled for a while, through several version upgrades I'm sure. |
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| 02:00.16 | p3nguin | This is a new deployment, so I am trying to make sure every bell rings and every whistle toots. |
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| 02:28.33 | p3nguin | Is there a plan to fix the voicemail bug? |
| 02:28.50 | p3nguin | How about the permissions on the voicemail files? Any setting for that? |
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| 03:11.17 | phix | Any recommendations for a SIP DECT phone? |
| 03:11.26 | phix | Or any ones to stay away from? |
| 03:12.07 | phix | I have been looking at the Snom M3 and Aastra 6757i CT |
| 03:12.40 | phix | Any one used these? |
| 03:13.20 | itsbroken | I've had bad experience with snom m3 |
| 03:14.01 | phix | oh, what happeend? |
| 03:14.05 | itsbroken | Haven't been able to figure out why they're reporting busy and not accepting calls |
| 03:14.13 | itsbroken | its random |
| 03:14.18 | itsbroken | client has 5 |
| 03:14.28 | itsbroken | sometimes one will do it... sometimes three |
| 03:15.02 | phix | and how are they configured? All the phones have seperate lines and dedicated transfer button or they share the same line? |
| 03:15.27 | phix | Does each phone make a seperate SIP connection? or does the basestation do this? |
| 03:15.29 | itsbroken | I've tried both.. but if you really wanna use those you gotta have just one line |
| 03:15.33 | itsbroken | on all |
| 03:15.34 | itsbroken | shared |
| 03:15.46 | phix | hmmmm that is not what I want |
| 03:15.55 | phix | They have a dedicated transfer button? |
| 03:16.07 | itsbroken | well look carefully at m3 because it says it can support like 5 on a base station |
| 03:16.07 | phix | hmmm I am looking at the Siemens C470 now |
| 03:16.18 | joako | Is there some other place besides packages.asterisk.org to get Asterisk RPM packages? |
| 03:16.26 | itsbroken | but only supports 3 phones at a time |
| 03:16.37 | itsbroken | if you want all 5 to ring you have to put the same extension on all of them |
| 03:16.46 | itsbroken | something crazy like that |
| 03:17.34 | joako | itsbroken: But can you say put 2 handsets on their own extension and the rest on the remaining? |
| 03:17.46 | phix | so you can't give each phone a seperate number / sip account? |
| 03:17.55 | itsbroken | joako: that may work |
| 03:18.18 | phix | I only want 2 handsets atm, at the most 4 |
| 03:18.34 | itsbroken | I would keep it wired if you can |
| 03:18.42 | phix | ok any other DECT SIP phones I should look at? |
| 03:18.46 | itsbroken | I haven't tried any wifi phones but I would try those before another dect |
| 03:18.58 | joako | I've used Aastra 9480 CT and it had similar limitations, the handset couldn't be a separate extension and I dunno if it would even do multiple handsets |
| 03:19.02 | phix | wifi phones are more expensive though :/ |
| 03:19.05 | joako | I haven't tried any wifi phone that worked well |
| 03:19.25 | phix | which wifi phones should i stay away from? |
| 03:19.38 | itsbroken | ask joako i haven't tried any |
| 03:19.49 | phix | what was directed at joako :) |
| 03:20.24 | phix | hmmmm CISCO SPA501G 8 line IP phone $120 :/ |
| 03:20.36 | phix | I have had very bad experience with CISCO / Linksys |
| 03:20.46 | joako | I just never found one that could work well. Something like stand 10 feet from the wifi AP and walk around and the call is breaking up |
| 03:20.49 | phix | I like my Snom 300 but it needs more buttons |
| 03:21.22 | phix | joako: u/alaw or gsm? |
| 03:22.23 | joako | WME/WMI & noack on off no different. Handoffs between AP however always seem to work great. |
| 03:22.36 | joako | I don't think any support GSM. Pretty sure I've tried G729 & Ulaw |
| 03:25.22 | phix | hmmmmm |
| 03:25.39 | itsbroken | what country are you in phix |
| 03:25.46 | phix | Australia |
| 03:26.01 | itsbroken | don't you want alaw then? |
| 03:26.11 | phix | I also need to get an ISDN phone card, although I don't think there is one that has been certified to work in AU |
| 03:26.27 | phix | Even though AU uses the EU standard it needs to be certified seperatly :/ |
| 03:26.37 | itsbroken | why don't you use sip? |
| 03:26.39 | phix | itsbroken: correct, alaw isused here |
| 03:27.13 | phix | itsbroken: I will use SIP for outgoing calls, but for incoming they want to use their existing numbers which don't port over to VoIP |
| 03:27.23 | phix | well there are no companies here that do that |
| 03:27.26 | itsbroken | >.< |
| 03:27.28 | phix | (That I know of) |
| 03:27.33 | itsbroken | moar research |
| 03:28.01 | phix | yeah I am in the middle of it :) so far nothing |
| 03:28.08 | phix | I am open to suggestions :) |
| 03:33.06 | phix | Still open to suggestions :) |
| 03:35.27 | phix | :( |
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| 03:56.20 | itsbroken | lol |
| 03:56.29 | itsbroken | gl =) |
| 04:04.32 | phix | hi |
| 04:04.57 | phix | hmmm hx8 card looks interesting |
| 04:05.14 | phix | any one set one of these ups? |
| 04:08.25 | phix | also i have a tdm24xxp card, and faxing doesnt work, hylafax cant train the modem plus the driver doesnt pick up the hardware timing thibgy |
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| 04:49.36 | p3nguin | This might be a clue about the voicemail thing: expected 7 but found 8 message(s) in box with max threshold of 100. I saw this go by on the CLI for another mail box. It is experiencing a similar problem to what mine is. |
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| 05:21.21 | Smirker | howdy. |
| 05:21.57 | Smirker | the Dial command. fairly straight forward and have been using it successfully. |
| 05:22.10 | Smirker | however one particular number, when i try to Dial it, all I can hear is blank. |
| 05:22.24 | Smirker | in the log it says Called xyz... |
| 05:22.45 | Smirker | usually after that it says "xyz is ringing" ... "bridging channels together" |
| 05:23.13 | Smirker | however it doesn't on this particular number. any using any other number works fine. however i can call the culprit number from any other phone perfectly. |
| 05:23.47 | Smirker | the only difference that i can tell, if that the number i am trying to call Answers straight-away without ringing, which makes we wonder if Asterisk isn't progressing because it hasn't heard the ring yet. |
| 05:23.47 | ChannelZ | is it pots? |
| 05:23.49 | Smirker | any ideas? |
| 05:24.10 | ChannelZ | or I should say, are you dialing out SIP or POTS or.. |
| 05:24.21 | Smirker | dialing out via SIP |
| 05:24.34 | ChannelZ | to a provider? |
| 05:25.07 | Smirker | i'm dialing to a call-centre. |
| 05:25.33 | ChannelZ | I mean you're going SIP to an ITSP of some sort I assume |
| 05:26.08 | Smirker | oh, yes. |
| 05:26.19 | ChannelZ | So barring any communications difficulties causing their SIP replies to disappear and not make their way back to you, if they are not reporting any call progress then it's something on their end |
| 05:26.25 | Smirker | my asterisk boxes are colocated in my providers DC and everything goes via private network. |
| 05:26.44 | Smirker | no problems with 100+ services, however just this one number is causing problems. |
| 05:27.11 | Smirker | ah okay. so i'll jump on the phone to my provider then |
| 05:27.15 | ChannelZ | You'll probably have to get them to debug it. Something bizarre about that one number and their T1 or whatever |
| 05:27.29 | Smirker | awesome. cheers |
| 05:27.39 | ChannelZ | sure good luck |
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| 05:29.55 | Smirker | ChannelZ: still around? i might throw one more question at you |
| 05:30.33 | Smirker | so we've got our server as i described, with 200 concurrent calls allowed. |
| 05:30.58 | Smirker | people here can call our DIDs fine, concurrently, etc. we've tried 20 times without a problem. |
| 05:31.22 | Smirker | at our SIP provider, they can call it fine too, over their network and over Telstra/Optus (australian telcos) |
| 05:31.57 | Smirker | however one of our other suppliers who supply our 1800/1300 numbers get issues. and they've told me the following: |
| 05:32.09 | Smirker | Iâve had a chat with one of our PABX specialist here and heâs mentioned that there is a configuration on either the DID and Phone server which needs to be configured the same, otherwise calls will occasionally fail. This config is called âclockingâ. |
| 05:32.23 | Smirker | does that sound reasonable/rational/logical? |
| 05:36.31 | SiNGLer | Smirker: returning to your previous question: it may be that on other end is a pbx, which does not send progress/ringing, so you do not hear it. that "other supplier"is SIP too? |
| 05:40.11 | phix | hi |
| 05:41.03 | SiNGLer | hi |
| 05:46.32 | Smirker | SiNGLer: it most likely is. however the calls don't get connected at all. |
| 05:46.52 | Smirker | when dialing from my SIP service, i get no status updates and i don't hear anything. |
| 05:47.05 | Smirker | when dialing from any other service, the call gets answered straight away and i hear artificial ringing. |
| 05:48.43 | *** join/#asterisk Ean (~Ean@unaffiliated/ean) |
| 05:48.55 | SiNGLer | oh, I thought that calls get connected, in this case I will not tell anything new :) ask your provider to debug or try looking at packet dump with tcpdump/wireshark |
| 05:49.22 | Smirker | ah yeah, emailing provider now |
| 05:49.40 | Smirker | i tried sip set debug on, and after I dial all I receive is a TRYING packet, and then nothing. |
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| 06:20.38 | Iiiak | hello |
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| 06:49.32 | schmidts | good morning |
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| 07:04.37 | pethkaqeni | morning all |
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| 07:33.42 | kaushal | hi |
| 07:33.44 | kaushal | Does it has Automated Dialing Feature like dialing 1000 and 1000 of phone numbers,Does it Support VoiceXML and What PRI Card is recommended for using FreeSwitch ? |
| 07:33.49 | kaushal | I mean asterisk |
| 07:36.24 | jkroon | no. |
| 07:36.37 | jkroon | at least not directly, i think. |
| 07:37.28 | jkroon | no voicexml support that I'm aware of, but you should be able to handle a large (few thousand) concurrent calls on decent hardware. |
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| 08:35.32 | nicola_pav | hello. i have installed free fax for asterisk on asterisk 1.4 with a pri card. I successfully send faxes. I wrote a shell script to convert the tiff to pdf and send it to my email. the FAXOPT variable remotestationid is always empty. i know its read only but is there any way around? |
| 08:36.05 | nicola_pav | i receive faxes successfully* |
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| 08:50.09 | coppice | nicola_pav: is the station ID set on the machine which is sending the FAX? |
| 08:53.57 | nicola_pav | coppice: how do i check it? |
| 08:54.18 | nicola_pav | its a samsung printer/fax |
| 08:54.44 | nicola_pav | coppice: to get this right. the fax sending machine should send this id |
| 08:54.57 | nicola_pav | it does not have to do anything with asterisk+ffa? |
| 08:55.00 | nicola_pav | is that right? |
| 08:55.12 | coppice | there should be a way to configure the station ID on any FAX machine. if you don't set one, it won't be sent |
| 08:55.37 | nicola_pav | coppice: i understand, i will try to check the fax machine. |
| 08:55.41 | nicola_pav | thanks a lot |
| 08:57.01 | *** join/#asterisk hensema (~erik@scrat.hensema.net) |
| 08:58.17 | hensema | hi all, I'm trying to decide wether it's worthwhile to upgrade from asterisk 1.4.x to asterisk 1.8.x |
| 08:58.38 | hensema | all timing related functions in asterisk 1.4 fail badly due to no zt_dummy driver in my kernel version |
| 08:59.04 | hensema | but I do need AgentCallBackLogin so I *think* asterisk 1.8 will force me to rewrite my entire dialplan in AEL |
| 09:01.09 | hensema | so my first question: will I be able to keep my extensions.conf *and* have AgentCallBackLogin functionality in asterisk 1.8? |
| 09:01.29 | hensema | and will asterisk 1.8 run better without zt_dummy module? |
| 09:01.54 | *** join/#asterisk Cadey (~x@62.84.178.106) |
| 09:02.11 | SiNGLer | zt_dummy is deprecated in 1.4, later versions use dahdi_dummy which is now integrated into dahdi module |
| 09:03.05 | SiNGLer | you can try upping dahdi version, then checking performance |
| 09:03.34 | Cadey | Hi guys, we are trying to use the AMI to work out who is calling a terminal but when the goes into a queue there is no link between the channel used to call into the queue and the channls the queue app creates to the terminals. All teh channels the queue app makes have blank CallerID's and so we cannot show the callerID to the user when there phone rings |
| 09:03.37 | SiNGLer | also callbacklogin can be replaced with some sort of auth + addqueuemember |
| 09:03.41 | Cadey | any suggestions? |
| 09:04.44 | Cadey | however the handsets do see the callerID which is strage as the AMI output has the callerID's as blank in the newchan message |
| 09:06.35 | hensema | SiNGLer: well, you're as vague as most documentation I can find.. "some sort of auth" |
| 09:06.44 | hensema | SiNGLer: that's why upgrading from 1.4 is *hard* |
| 09:16.09 | kaldemar | hensema: i don't see why you would be forced to use AEL. you'll get a pretty good picture of required extensions.conf and other changes by taking a look at UPGRADE*.txt in a 1.8 source package. |
| 09:16.55 | kaldemar | hensema: but for the basic question you were asking, it will be worthwhile when 1.4 gets its end-of-life and will no longer receive bug fixes. |
| 09:17.36 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 09:17.38 | hensema | kaldemar: ok, I'll try to figure out the 1.8 equivalent of AgentCallbackLogin |
| 09:17.49 | *** join/#asterisk Ean (~Ean@unaffiliated/ean) |
| 09:18.44 | kaldemar | hensema: and if you update to 1.8 you'll be more likely to get help with possible issues when you're using what everyone else is. |
| 09:21.59 | kaldemar | hensema: UPGRADE-1.4.txt states that AgentCallbackLogin is deprecated since the entire function it provides can be done with dialplan logic and encourages to look at http://svn.digium.com/svn/asterisk/tags/1.4.40.1/doc/queues-with-callback-members.txt |
| 09:22.43 | hensema | kaldemar: yes, but that's AEL and my entire dialplan is in extensions.conf |
| 09:22.56 | hensema | also it's a *lot* of code to replace a few configuration lines |
| 09:23.15 | hensema | so I'm a bit at a loss here |
| 09:24.23 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 09:24.43 | kaldemar | you can always use both extensions.conf and extensions.ael if you don't wish to rewrite it. |
| 09:25.08 | *** join/#asterisk atiti (~atiti@192.38.94.172) |
| 09:27.02 | kaldemar | there's also a conf2ael utility in asterisk. |
| 09:27.44 | SiNGLer | also you can rewrite ael code into conf (afaik aelparse can provide that) |
| 09:31.27 | gerhard7 | hi, anyone running skype for asterisk? |
| 09:35.12 | *** join/#asterisk wonderworld (~ww@port-92-201-55-14.dynamic.qsc.de) |
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| 09:40.34 | hensema | gerhard7: I am, though it's broken at the moment |
| 09:42.43 | Cadey | conf2ael utility <-- but doesnt asterisk still convert all AEL back to conf format anyway to actualy action the dial plan? |
| 09:43.07 | Cadey | I know it did in 1.6 anyway :S |
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| 10:55.03 | *** join/#asterisk vampi-the-frog (~zyx@mail.cmbtravel.ro) |
| 10:55.19 | vampi-the-frog | hey guys i just installed asteriskNOW and i've no idea where to start, does it have a web interface or something? |
| 10:55.44 | vampi-the-frog | or rather - any documents referring to it? |
| 10:55.49 | vampi-the-frog | (to asteriskNOW) |
| 10:56.33 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 11:03.55 | vampi-the-frog | alright, i found it at :8088, but it's asking me for a user and password |
| 11:04.18 | vampi-the-frog | and all i have is root |
| 11:04.43 | kaldemar | vampi-the-frog: http://www.asterisk.org/AsteriskNOW-1.5-QuickStart |
| 11:05.08 | vampi-the-frog | ty |
| 11:06.14 | vampi-the-frog | dang |
| 11:06.25 | vampi-the-frog | i chose the Asterisk GUI boot prompt :/ |
| 11:06.30 | vampi-the-frog | is FreePBX recommended for beginners? |
| 11:06.59 | Cadey | well FreePBX is kinder self contained |
| 11:07.14 | Cadey | if you want to learn about asterisk a vanila install is a good way to learn |
| 11:07.36 | Cadey | FreePBX is good if you need a working PBX with little to no nowledge of how it actualy works |
| 11:08.10 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 11:08.17 | vampi-the-frog | yeah that |
| 11:08.21 | vampi-the-frog | little to no knowledge :) |
| 11:08.27 | Cadey | :) |
| 11:08.35 | vampi-the-frog | i'll learn in time but right now we need a working install quick |
| 11:08.48 | Cadey | yeah freePBX would be OK for that then :) |
| 11:08.57 | vampi-the-frog | ok so i guess i'll reinstall with freepbx? or is there a simpler way to switch? some yum command? |
| 11:09.11 | Cadey | but when you want to learn more about asterisk and how it actualy works idinstall and play aroud with a vanila asterisk install |
| 11:09.54 | Cadey | im not sure about that vampi because FreePBX will not use exactly the same source as a normal build of asterisk |
| 11:10.10 | vampi-the-frog | ah well |
| 11:10.12 | vampi-the-frog | reinstall it is |
| 11:10.13 | vampi-the-frog | thanks |
| 11:10.15 | Cadey | they have a customised version of it I belive |
| 11:10.37 | vampi-the-frog | just to make sure, this is what i mean: http://www.asterisk.org/images/AsteriskNOW-1.7.0_Boot_Menu.png |
| 11:10.39 | kaldemar | freepbx is just a GUI for asterisk. it does not include asterisk. |
| 11:10.45 | vampi-the-frog | i typed 4 |
| 11:11.00 | vampi-the-frog | kaldemar: so can i switch from Asterisk GUI to FreePBX? |
| 11:11.04 | vampi-the-frog | without reinstalling |
| 11:11.56 | Cadey | kaldemar : doesnt freePBX have is own installer and version asterisk to go with it? |
| 11:12.19 | kaldemar | vampi-the-frog: i guess there's no technical obstacles, but i have never used asterisknow so i don't know. btw, there are also #asterisknow and #freepbx channels. this channel is more concentrated on pure asterisk. |
| 11:12.27 | Cadey | sorry vampi, I guess I was wrong then :( |
| 11:13.21 | kaldemar | Cadey: not that i know of. there are quite a few scripts flying around that install freepbx, asterisk, dahdi though. |
| 11:16.07 | vampi-the-frog | oh |
| 11:16.09 | vampi-the-frog | ty |
| 11:21.31 | *** join/#asterisk bratner (~bratner@95.211.21.37) |
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| 11:36.28 | Lantizia | Hey do we've got a few phone systems out there (one per customer) that are asterisk/freepbx combinations... were thinking of becoming a virtual ITSP (i.e. resell a major ITSP's services - but we only get one master account off them and need to terminate it on our network first before our customers connect to us rather than directly to them)... now I'd like to use asterisk for the endpoint but obviously freepbx aint gonna cut it but i need some k |
| 11:36.28 | Lantizia | ind of interface for our engineers who are not as familiar with freepbx... so far I've been told about some called eConsole - any others people know of? |
| 11:37.20 | Lantizia | s/as familiar with freepbx/as familiar with asterisk |
| 11:51.49 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
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| 12:17.49 | *** join/#asterisk ManWithNoName (~ManWithNo@200.242.28.231) |
| 12:21.07 | ManWithNoName | Hello. I have an * server connected to a Siemens HiPath 4000 via E1 PRI. Everything is working ok, except when I try to enter a meetme conference room using a siemens digital phone. In that situation the * server does not recognie the digits I press to identify the room. When I use an analog line it works ok. |
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| 12:46.14 | hensema | I must be missing something here: |
| 12:46.29 | hensema | in previous versions you could do something like this in extensions.conf: |
| 12:46.31 | hensema | [globals] |
| 12:46.34 | hensema | FOO=123 |
| 12:46.37 | hensema | [mycontext] |
| 12:46.51 | hensema | exten => ${FOO},Answer() |
| 12:47.07 | leifmadsen | well first off you're missing a priority number |
| 12:47.13 | leifmadsen | that would have never worked |
| 12:47.18 | leifmadsen | (in any version) |
| 12:47.32 | leifmadsen | exten => ${FOO},1,Answer() |
| 12:47.42 | hensema | you're right of course |
| 12:47.59 | hensema | but the problem here is the global definition of ${FOO} |
| 12:48.21 | hensema | exten => ${GLOBAL(FOO)},1,Answer() <-- doesn't work |
| 12:48.47 | hensema | exten 1,1,DoSomething(${GLOBAL(FOO)} <-- may work, haven't tested it yet |
| 12:49.00 | ManWithNoName | I'm sorry, I lost my connection! I'm trying to access a Meetme room through a siemens digital phone and the * server does not recognize the digits I enter to identify the room. If I use an analog line it worsk fine! The * and the Hipath are connected via E1 PRI |
| 12:51.54 | hensema | any clue on how to reference a global variable in an extension? |
| 12:52.03 | leifmadsen | hensema: there must be a typo, because I just tested it here on trunk, and it works fine |
| 12:52.17 | hensema | hmmm weird |
| 12:52.17 | leifmadsen | [globals] |
| 12:52.19 | leifmadsen | FOO=123 |
| 12:52.22 | leifmadsen | [default] |
| 12:52.28 | leifmadsen | exten => ${FOO},1,NoOp() |
| 12:52.36 | leifmadsen | works fine, no errors |
| 12:52.48 | *** join/#asterisk sourcode (~code@ppp-58-11-74-190.revip2.asianet.co.th) |
| 12:55.00 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 12:55.16 | hensema | in in asterisk 1.8.3: |
| 12:55.20 | hensema | [globals] |
| 12:55.22 | hensema | NUM_HOSTINGXS=243249177 |
| 12:55.27 | hensema | [default-incoming] |
| 12:55.31 | hensema | exten => ${NUM_HOSTINGXS},1,Goto(hostingxs-menu,s,1) |
| 12:55.35 | hensema | results in: |
| 12:55.40 | hensema | [Apr 6 14:54:33] ERROR[11744]: pbx.c:8090 ast_add_extension2_lockopt: You have to be kidding-- add exten '' to context default-incoming? Figure out a name and call me back. Action ignored. |
| 12:58.39 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:01.13 | hensema | ah, found it |
| 13:01.35 | hensema | I got two [globals] sections and defined the vars in the second |
| 13:01.38 | hensema | d'oh |
| 13:02.37 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 13:03.32 | *** join/#asterisk pethkaqeni (d5cf2336@gateway/web/freenode/ip.213.207.35.54) |
| 13:03.37 | pethkaqeni | hi all |
| 13:06.00 | *** join/#asterisk kamikazemicrowav (~kamikazem@63.246.22.126) |
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| 13:40.39 | zkn | hello, help me remember if there is a function or variable that I can use to check what context the call or any other dialplan process started from ? or do I need set these markers as variables in the dialplan manually if i need to be able to call out this information ? |
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| 13:42.59 | saxa | zkn: i'm not sure, but probably you need to use the Verbose() in each context |
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| 13:43.59 | zkn | well, i know there is ${CONTEXT} variable built in, i remember there being something like ${FROMCONTEXT} too... just not sure... |
| 13:44.05 | _Corey_ | zkn: Make sure verbose logging is turned on and look at the first line when the call enters your dial plan |
| 13:44.22 | _Corey_ | It will look something like this: |
| 13:44.23 | _Corey_ | VERBOSE[25580] pbx.c: -- Executing [555@sip:1] Ringing("SIP/2010-0000078b", "") in new stack |
| 13:44.44 | _Corey_ | in this example 'sip' is the context and '1' is the priority |
| 13:44.44 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 13:45.03 | Yedidya | New Question. NOTICE[19839] chan_iax2.c: Rejected connect attempt from x.x.x.x, requested/capability 0x2/0x783 incompatible with our capability 0x8. I think 0x8 fits within 0x783 |
| 13:45.16 | Yedidya | I think 0x8 fits within 0x783 |
| 13:46.04 | Yedidya | does it? if yes why reject? |
| 13:47.18 | kaldemar | zkn: CONTEXT has the current context in it, so it can change. you could either set it to a variable in the dialplan or in a channel configuration file with a setvar parameter. |
| 13:48.03 | zkn | kaldemar, yes that's what I thought when i mentioned "set these markers as variables in the dialplan manually" |
| 13:52.52 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 13:54.15 | kaldemar | zkn: for SIP, you could use ${SIPPEER(context)}. |
| 13:56.26 | *** join/#asterisk Ean (~Ean@unaffiliated/ean) |
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| 13:58.33 | jero | I want to implement a "page-then-transfer" feature here (page callee to inform him of incoming call, then transfer the actual call). What would be the proper way to implement this ? |
| 14:00.36 | pabelanger | Yedidya: what version of Asterisk you using? |
| 14:05.07 | Yedidya | pabelanger: 1.6.2.17 |
| 14:05.10 | *** join/#asterisk Greek-Boy (~Greek-Boy@41.191.92.29) |
| 14:06.02 | Yedidya | does this make sence? |
| 14:06.03 | Yedidya | > requested format = gsm, |
| 14:06.06 | Yedidya | <PROTECTED> |
| 14:06.07 | Yedidya | <PROTECTED> |
| 14:06.09 | Yedidya | <PROTECTED> |
| 14:06.10 | Yedidya | <PROTECTED> |
| 14:06.32 | Yedidya | if I have proirity then why is it not alaw? |
| 14:06.59 | pabelanger | ~pb |
| 14:07.00 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:07.13 | pabelanger | Yedidya: ^ pastebin the full output |
| 14:07.21 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:07.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:07.25 | pabelanger | ~collectdebug |
| 14:07.25 | infobot | it has been said that collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 14:08.27 | Yedidya | ok, it may take time becuase the pstn to voip provider use an array of servers and only some have this issue. |
| 14:11.28 | Yedidya | pabelanger: what level of debug whould you like to see? |
| 14:12.04 | pabelanger | Yedidya: all the required steps are on the wiki page |
| 14:18.05 | *** join/#asterisk minaguib (~mina@modemcable098.129-202-24.mc.videotron.ca) |
| 14:18.40 | minaguib | Hey guys. If I'm looking at a phone's specs and it says 'PoE required, AC Adapter optional', does that mean I can use an AC adapter instead of PoE ? or by ac adapter they mean a PoE power injector - not included ? |
| 14:19.08 | *** join/#asterisk Zylogue (~Zylogue@c-71-194-90-188.hsd1.in.comcast.net) |
| 14:21.37 | Yedidya | pabelanger: http://pastebin.com/jh81sGzh |
| 14:21.59 | zkn | how could I see what information an inbound SIP call consists of ? e.g. using Verbose + variables to output that info to console to get an idea what I could later use in the dialplan during the callflow .. |
| 14:22.58 | Zylogue | I'm a bit 'new' to this idea for Asterisk and know nothing about it, other than it is a voip solution. Where can I find answers to questions like: what hardware do I need to make user of an Asterisk server? Where do I get the phone number from? Can I port an existing number? What is SIP and PSTN? Thanks |
| 14:23.02 | longword | mina: Not something we could possibly answer for a product we don't sell |
| 14:23.05 | pabelanger | zkn: *CLI> core show application DumpChan() |
| 14:23.52 | longword | IME many phones have a DC socket and will happily live off that plus plain old fashoned non-Po Ethernet |
| 14:25.11 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 14:26.30 | ruben23 | hi guys i have added on my asterisk some sip phone extensions and i can see it on like one week then suddenly when i sip show peers its empty alredy any idea, please |
| 14:27.56 | zkn | pabelanger: thanks, i discovered now I had disabled app_dumpchan.so - will look into it now |
| 14:28.07 | pabelanger | Yedidya: what am I looking at? What is the issue |
| 14:30.09 | Yedidya | oh, sorry, like i said, if my pref codec is alaw and the other party supports it and the priority is mine then why is the actual codec gsm? |
| 14:32.26 | Yedidya | pabelanger: oh, sorry, like i said, if my pref codec is alaw and the other party supports it and the priority is mine then why is the actual codec gsm? |
| 14:33.03 | pabelanger | Yedidya: looks like it is using alaw to me |
| 14:33.05 | pabelanger | [Apr 6 15:25:44] DEBUG[18377] channel.c: Set channel IAX2/77.240.48.140:4572-10764 to write format alaw |
| 14:34.06 | pabelanger | Yedidya: also, do you have .alaw prompts for everything? To avoid transcoding |
| 14:34.08 | Yedidya | I see, what then is the line stating 47. " > actual format = gsm," ? |
| 14:34.32 | Yedidya | pabelanger: yes, propts are all alaw. |
| 14:36.38 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 14:37.32 | Yedidya | pabelanger: I see, what then is the line stating 47. " > actual format = gsm," ? |
| 14:38.17 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 14:38.17 | pabelanger | Yedidya: is this an inbound or outbound call |
| 14:38.23 | pabelanger | first part of debug log is missing |
| 14:38.48 | Yedidya | pabelanger: inbound |
| 14:39.10 | pabelanger | Yedidya: Are you using codecpriority= ? |
| 14:39.16 | pabelanger | in iax.conf |
| 14:39.37 | Yedidya | pabelanger: codecpriority= ? no. |
| 14:39.40 | pabelanger | Yedidya: PB your iax.conf, besure to remove any password |
| 14:39.49 | pabelanger | s/password/passwords |
| 14:40.46 | Yedidya | pabelanger: this is a freebpx configured system, will have to concatenate - stand by.... |
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| 14:48.41 | Yedidya | pabelanger: http://pastebin.com/c3gj0cxB |
| 14:49.35 | pabelanger | Yedidya: is this one file? or did you combined them for pastebin? |
| 14:50.03 | Yedidya | combined |
| 14:50.10 | Yedidya | pabelanger: combined |
| 14:53.37 | pabelanger | Yedidya: makes it hard to read, but you inbound IAX call is UNAUTHENTICATED, meaning it is not using a context defined in iax.conf. So, it will get your default settings in [general]. Make sure your disallow / allow order is correct. It is too hard to follow the FreePBX logic. |
| 14:53.57 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 14:54.11 | pabelanger | you may need to try #freepbx for their software |
| 14:55.12 | *** join/#asterisk mocker (~mocker@206.55.118.83) |
| 14:55.14 | Aut0ExeC | hi guys.. whats the benefits of using switchvox vs asterisk? |
| 14:55.31 | Aut0ExeC | nicer gui interface? |
| 14:55.46 | Aut0ExeC | and boxes that are already set up with hardware? |
| 14:56.02 | malcolmd | Aut0ExeC: one is a phone system in a box (switchvox), the other is a toolkit for building telephony applications (asterisk) |
| 14:57.18 | Aut0ExeC | malcolmd: so essentially you can get asterisk to do everything that switchvox can do |
| 14:57.18 | Yedidya | pabelanger: My error, I combined also iaxprov.con which makes up the bulk of that paste. here is the correct one http://pastebin.com/L6J2rapH |
| 14:57.21 | Aut0ExeC | ? |
| 14:59.24 | pabelanger | Aut0ExeC: with enough work, yes |
| 14:59.42 | Aut0ExeC | pabelanger: ok.. I just need a solid box with a solid digium card yes? |
| 14:59.44 | malcolmd | "can get" is a rather loose term....given the time, the inclination, and the skill, you could use asterisk, a number of other ots OSS components, and develop a switchvox like thing. think of astersk as all of the tools that you need to build a house (hammer, nails, wood) while switchvox is a 2-level home with a garage and a kitchen |
| 14:59.46 | pabelanger | switchvox is a product. |
| 15:00.16 | Aut0ExeC | malcolmd: lol nice way of putting it ... lol thanks |
| 15:00.54 | Aut0ExeC | is everything gui in switchvox? |
| 15:00.57 | malcolmd | yes |
| 15:01.01 | Aut0ExeC | ok |
| 15:01.05 | malcolmd | aside of the api http://developers.digium.com/switchvox |
| 15:01.11 | Aut0ExeC | k |
| 15:01.23 | malcolmd | there's no "console" access for switchvox |
| 15:01.26 | Aut0ExeC | sounds like a different beast altogether |
| 15:01.31 | Aut0ExeC | if ur used to configuring with asterisk |
| 15:01.34 | Aut0ExeC | :| |
| 15:01.39 | Aut0ExeC | but i get ur analogy |
| 15:02.34 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:03.57 | pabelanger | Aut0ExeC: depends on what you need to do honestly |
| 15:07.06 | Aut0ExeC | pabelanger: ok thanks man |
| 15:07.43 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 15:08.16 | Aut0ExeC | i figure the basic switchvox seems to cost like 3000+ .. a nice digium card cost like 2000+ excluding the pc.. so might as well get the vox if your going big |
| 15:08.29 | Aut0ExeC | ya dig? |
| 15:08.47 | pabelanger | which cards do you need? |
| 15:09.49 | leifmadsen | A Switchvox system for about $3000 isn't too bad since I've found recently it's nearly impossible to build a custom Asterisk system for customers without charging them about $3000. |
| 15:10.04 | leifmadsen | at least not to make it worth my time :) |
| 15:10.09 | Aut0ExeC | leifmadsen: yeah exactly |
| 15:10.54 | leifmadsen | neither is better than the other, but when people say switchvox is too expensive, they haven't tried hiring a consultant to build them a system before :) |
| 15:11.31 | Aut0ExeC | pabelanger: i was looking at our company.. right now we have a T1 with I think 2 24 port cards |
| 15:11.32 | leifmadsen | the latter is easier to customize though and you just get what you want, and not all the bells and whistles you may not need |
| 15:11.58 | leifmadsen | a 24 port card sounds like a TDM2400p |
| 15:12.02 | leifmadsen | (analog lines) |
| 15:12.09 | Aut0ExeC | leifmadsen: yeah... |
| 15:12.18 | Aut0ExeC | you'd rather build an asterisk server? |
| 15:12.20 | leifmadsen | unless you mean single span card (which can handle 24 channels on a single T1) |
| 15:12.28 | leifmadsen | well I build asterisk servers all the time |
| 15:12.44 | leifmadsen | I just deployed for 2 companies recently, and will be creating a distributed system for a union |
| 15:12.47 | Aut0ExeC | i mean fully featured to customers specs? vs switchvox |
| 15:13.11 | Aut0ExeC | i guess if u build 1, u've built them all huh |
| 15:13.21 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 15:13.23 | leifmadsen | I always build to customers specs. It wouldn't make sense to build something out-of-spec |
| 15:13.24 | krion | hi |
| 15:13.35 | Aut0ExeC | leifmadsen: u do that for a living? |
| 15:13.43 | krion | i use asterisk with a sip trunk in order to get my call to extern |
| 15:13.48 | leifmadsen | Aut0ExeC: not true -- I haven't yet run into a system that is exactly the same as another I've built |
| 15:13.53 | leifmadsen | Aut0ExeC: you could say that |
| 15:14.02 | Aut0ExeC | leifmadsen: thats kewl man |
| 15:14.04 | Aut0ExeC | :) |
| 15:14.08 | krion | the things is my sip trunk provider says my From field is incorrect |
| 15:14.11 | krion | http://pastebin.com/p9QZRu8K |
| 15:14.13 | leifmadsen | ~leifmadsen |
| 15:14.13 | infobot | you are, like, blitzrage |
| 15:14.18 | leifmadsen | heck ya I am |
| 15:14.28 | krion | does it look incorrect to you ? |
| 15:14.51 | leifmadsen | krion: that is the To field |
| 15:15.11 | leifmadsen | now I see the From field, but we can't tell if it is incorrect or not |
| 15:15.15 | pabelanger | krion: you best to pb a full debug log of you issue, not just select lines |
| 15:15.25 | leifmadsen | it's obviously not what your provider is expecting |
| 15:15.51 | krion | pabelanger: ok i can have that |
| 15:16.05 | krion | leifmadsen: obviously... |
| 15:17.30 | leifmadsen | krion: they probably expect the From to contain your account name and not 'asterisk' |
| 15:18.01 | krion | leifmadsen: the things is i want it to be anonymous, but as i read the rfc is not the proper way to do it |
| 15:18.58 | leifmadsen | ok |
| 15:21.10 | *** join/#asterisk mocker (~mocker@206.55.118.83) |
| 15:21.51 | mocker | Looking for advice on devices to take a DS3 and convert the signal to SIP that can be routed to Asterisk. Anyone have suggestions on potential devices? |
| 15:22.35 | wonderworld | out of curiosity: what is DS3 ? |
| 15:23.10 | mocker | wonderworld: Like a PRI, but bigger. :P |
| 15:23.28 | wonderworld | ahh kk :) |
| 15:23.36 | wonderworld | how many channels? |
| 15:23.43 | mocker | 644 |
| 15:23.53 | wonderworld | now THATS a party line |
| 15:25.32 | krion | leifmadsen: told me that i should use sip:phonenumber and something about privacy name |
| 15:25.50 | krion | i got 3 different trunk provider, they all working differently... |
| 15:25.56 | krion | headache |
| 15:26.14 | *** join/#asterisk megalomano (~kvirc@nggw-of.alocomm.com) |
| 15:26.15 | *** join/#asterisk daxt (~daxt@112.135.91.64) |
| 15:26.47 | daxt | guys do you know a sip service provider who offers amr-nb codec support ? |
| 15:27.10 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:27.28 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:28.21 | megalomano | hi , someone knows how to set incoming calls to customer-card into a2billing ,,, thanks |
| 15:28.24 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 15:29.15 | *** join/#asterisk wesphillips (~wphillips@137.237.233.124) |
| 15:29.33 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:33.18 | vampi-the-frog | hey any idea why *CLI would be saying No such command 'core'? |
| 15:35.16 | d_preston215 | Does any one use Cisco 7940/60s and have come across the issue of time and date randomly disappearing? |
| 15:36.12 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 15:38.31 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 15:40.02 | *** join/#asterisk oej (~olle@94.127.50.104) |
| 15:42.23 | lost_soul | vampi-the-frog: how are you loading asterisk? |
| 15:42.30 | vampi-the-frog | asterisk -r |
| 15:42.31 | vampi-the-frog | as roo |
| 15:42.31 | vampi-the-frog | t |
| 15:43.21 | vampi-the-frog | this is the AsteriskNOW distro, 1.7.1, 64-bit |
| 15:43.21 | lost_soul | ah, so not through a script then.. found a thread someone had similar trouble and starting it using "asterisk -c" worked while the starting via rc script wouldn't |
| 15:43.35 | vampi-the-frog | oh wait |
| 15:43.43 | vampi-the-frog | i thought you meant how i access the CLI |
| 15:43.47 | vampi-the-frog | but yeah it's in rc |
| 15:43.56 | lost_soul | http://lists.digium.com/pipermail/asterisk-users/2009-February/226829.html |
| 15:44.13 | lost_soul | maybe as a test.. stop it completely.. start it via cli.. and see |
| 15:44.21 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 15:46.42 | zkn | is there an equivalent to SIPCALLID with IAX2? |
| 15:47.16 | vampi-the-frog | lost_soul: thanks, i'll try |
| 15:48.22 | vampi-the-frog | well actually |
| 15:48.37 | vampi-the-frog | 'core show help core' works. just when typing 'core' plainly it complains about it |
| 15:52.58 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 15:56.19 | *** join/#asterisk Goshen (~Goshen@c-174-52-7-122.hsd1.ut.comcast.net) |
| 15:56.32 | Goshen | Good morning |
| 15:58.00 | Aut0ExeC | Goshen: safer to just say hi |
| 15:58.01 | Aut0ExeC | :| |
| 15:58.07 | Aut0ExeC | afternoon in some places here |
| 15:58.27 | Goshen | Very true :) |
| 15:58.42 | Goshen | But just in case someone forgot to tell you Good morning ;) |
| 15:58.43 | Aut0ExeC | and u dont want to give your location away.. ur obviously eastern |
| 15:58.50 | Aut0ExeC | ok i see |
| 15:59.53 | *** join/#asterisk minaguib (~mina@modemcable098.129-202-24.mc.videotron.ca) |
| 16:01.59 | Goshen | my state and IP address are not hidden anyhow so no worries there |
| 16:06.44 | Aut0ExeC | k |
| 16:07.19 | mocker | ~ds3 |
| 16:07.19 | infobot | it has been said that ds3 is 28 T1 channels, or 672 individual B channels. |
| 16:10.41 | mocker | Looking for advice on devices to take a DS3 and convert the signal to SIP that can be routed to Asterisk. Anyone have suggestions on potential devices? |
| 16:10.43 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 16:11.11 | mocker | I've read about the Dialogic IMG 1010 and the Lucent Max TNT |
| 16:21.08 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 16:23.30 | *** join/#asterisk sourcode (~code@ppp-58-11-74-190.revip2.asianet.co.th) |
| 16:38.34 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 16:39.33 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:40.44 | zkn | i need advice on function CUT. i need to CUT out everything in SIPURI execept the IP address that comes after character "@", how do I do that? |
| 16:47.14 | *** join/#asterisk ttpears (~ttpears@ip24-250-214-113.ga.at.cox.net) |
| 16:52.24 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 16:52.38 | leifmadsen | zkn: |
| 16:52.43 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 16:52.45 | leifmadsen | exten => start,1,NoOp() |
| 16:52.56 | leifmadsen | same => n,Set(SIPURI=someone@someplace.com) |
| 16:53.09 | leifmadsen | same => n,Set(Result=${CUT(${SIPURI},@,1)}) |
| 16:53.25 | leifmadsen | same => n,NoOp(${Result)}) |
| 16:53.32 | leifmadsen | output of NoOp(): someone |
| 16:54.01 | zkn | leifmadsen: i want the result to be "someplace.com" |
| 16:54.16 | zkn | i have managed to get someone |
| 16:55.27 | ttpears | In AEL, is doing "if ( ${DIRECTION} == FOO) the same as doing if ( ${DIRECTION} = FOO)" ? |
| 16:59.48 | leifmadsen | zkn: change 1 to 2 |
| 16:59.48 | leifmadsen | you want the 2nd field then |
| 16:59.57 | leifmadsen | @ is the separator, and anything after the first @ is the second field |
| 17:00.06 | leifmadsen | field1@field2@field3@field4 |
| 17:00.14 | leifmadsen | that's how CUT() works |
| 17:01.53 | zkn | correct you are... i was trying -1 and 1- .. geesh :) |
| 17:02.04 | leifmadsen | heh |
| 17:02.12 | zkn | thank you! |
| 17:08.55 | jaytee | that's why everyone should purchase a copy of Asterisk: The Definitive Guide |
| 17:09.23 | jaytee | and the Asterisk Cookbook when it's available |
| 17:09.40 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 17:10.13 | leifmadsen | jaytee: soooooooooooooooooooon |
| 17:10.32 | leifmadsen | if you buy it now you'll get a copy as soon as it ships :) |
| 17:10.46 | leifmadsen | before the end of April afaik |
| 17:13.30 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 17:14.41 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 17:16.32 | zkn | i'm seriously considering buying The Definitive Guide... i was shocked to find that oreilly did not have it online anymore :) |
| 17:17.48 | leifmadsen | :) |
| 17:17.53 | leifmadsen | they removed it from the OFPS site it seems |
| 17:18.13 | leifmadsen | we have to work on a way of getting it put back up once all the changes they made outside of DocBook get put back into the repo. |
| 17:20.06 | leifmadsen | when O'Reilly gets the docbook source it pulls it out of subversion and puts it into another application for page layouts and all sorts of things, then all the changes have to be put back into DocBook manually. |
| 17:20.20 | leifmadsen | which as you can imagine is a lot of work |
| 17:20.31 | leifmadsen | and if you can't, go play with DocBook for a while :) |
| 17:29.32 | zkn | is it possible to used ExecIf inside ExecIf ? |
| 17:29.42 | zkn | s/used/use |
| 17:32.17 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v005-204.mobile.uci.edu) |
| 17:41.28 | Corydon76-home | zkn: No |
| 17:42.05 | Corydon76-home | You can use ${IF()} within ExecIf, however |
| 17:42.54 | *** join/#asterisk Godfather_ (~estanteri@89.131.93.52) |
| 17:43.18 | Godfather_ | hi |
| 17:43.23 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 17:43.38 | *** join/#asterisk sourcode (~code@58.11.74.190) |
| 17:44.14 | Godfather_ | when you have a register statement in sip.conf ( register => user:pw@ip/ddi ) how asterisk knows wich context has to entry? |
| 17:47.43 | leifmadsen | Godfather_: you define a [peer_definition] for that part |
| 17:47.51 | leifmadsen | the registration is ONLY to tell the other end where you exist on the network |
| 17:47.58 | leifmadsen | it has nothing to do with inbound authentication |
| 17:52.03 | Godfather_ | leifmadsen, then, you define a peer with fromdomain and fromuser, and you set the context here, and the fromuser="xxx "secret="yyy" should be the same as the statement register => xxx:yyy@fromdomain/ddi no? |
| 17:53.40 | Godfather_ | leifmadsen, i dont understand why you define a "peer" instead of a friend, cause you are able to receive incoming calls from it |
| 17:58.18 | Godfather_ | leifmadsen, http://pastebin.com/Q0CbSBv6 <- see that example, i dont know if its correct |
| 18:01.24 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 18:03.52 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 18:05.37 | mocker | leifmadsen: Any ebook+print combos? |
| 18:06.24 | leifmadsen | mocker: yep -- check the O'Reilly and Amazon sites |
| 18:06.28 | leifmadsen | they build stuff like that all the time |
| 18:07.04 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 18:07.08 | Godfather_ | leifmadsen, did you understand my question? i dont know where to place the incmoing calls form a register statement |
| 18:08.56 | mocker | leifmadsen: Ahh, ebook isn't available yet so I'll have to wait. |
| 18:10.51 | leifmadsen | mocker: ya they are still working on finishing the layout (page breaks etc) and getting it printed |
| 18:11.02 | leifmadsen | once physical books start shipping then the the ebook will likely become available |
| 18:11.12 | leifmadsen | Godfather_: sorry, working on other things and can't take a look |
| 18:11.22 | bent_screwdriver | trying to log failed faxes. any idea why this wouldn't work: exten => failed,n,Log(ERROR,FaxFailed) |
| 18:14.33 | bent_screwdriver | http://pastebin.com/CxH6SKJa |
| 18:14.56 | Godfather_ | leifmadsen, no prob |
| 18:15.05 | citywok | afaik, there is no log() command... you have to NoOp() |
| 18:15.21 | Godfather_ | when you have time take a look pls, its very easy example |
| 18:16.26 | bent_screwdriver | LOG() writes to the log "Send arbitrary text to a selected log level" |
| 18:16.51 | bent_screwdriver | Asterisk 1.6.2.9-2 |
| 18:24.12 | leifmadsen | bent_screwdriver: is LOG() a function or application? |
| 18:24.22 | bent_screwdriver | application |
| 18:25.23 | mocker | Looking for advice on devices to take a DS3 and convert the signal to SIP that can be routed to Asterisk. Anyone have suggestions on potential devices? |
| 18:26.48 | *** join/#asterisk NirS (~NirS@bzq-82-80-100-10.dsl.bezeqint.net) |
| 18:28.00 | bent_screwdriver | @leifmadsen: http://www.voip-info.org/wiki/view/Asterisk+cmd+Log |
| 18:31.47 | leifmadsen | shrugs |
| 18:32.50 | bent_screwdriver | @leifmadsen: weird...maybe its an old app that they left in but haven't updated. I really don't see too many people using it. I'll just use system to echo out to a fax log file. thanks |
| 18:41.26 | *** join/#asterisk tyrrexrrg (~roger@200.71.44.38) |
| 18:42.23 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 18:45.49 | ttpears | Anyone know if "==" is equivalent to "=" in variable reference syntax, like it is in expression syntax? |
| 18:47.20 | *** join/#asterisk mawhii (~mawhii@170.220.119.70.cfl.res.rr.com) |
| 18:53.47 | _Corey_ | mocker: Most people I know who still have DS3s are using Cisco SIP gateways |
| 18:54.03 | _Corey_ | I don't think Adtran has one |
| 18:54.04 | Qwell | ttpears: define "variable reference syntax" |
| 18:56.07 | *** join/#asterisk jkroon (~jkroon@dsl-241-249-95.telkomadsl.co.za) |
| 18:58.39 | lost_soul | I understand this is more a question of preference and best practice.. But I was wondering what other services would be considered ok to have on the same system as asterisk as it needs to have ports open to the outside world anyways.. I'm considering running a web server and perhaps a mail server on it. |
| 18:58.54 | lost_soul | just trying to get opinions on it, thanks |
| 18:59.02 | Qwell | best practice, none |
| 18:59.13 | lost_soul | lol, yea.. I know |
| 18:59.25 | Qwell | reality, whatever the hell you want. if it gets hacked, so does your PBX. |
| 18:59.28 | lost_soul | but I can't afford one service per system unfortunately |
| 18:59.54 | Qwell | How much is it worth to you, to prevent $100,000 in calls? |
| 19:00.16 | mocker | _Corey_: Know any model numbers? |
| 19:00.27 | lost_soul | Qwell: my service would just get shut off.. It wouldn't rack up that much of a bill |
| 19:00.38 | lost_soul | admittedly I don't want that to happen either though |
| 19:00.45 | mocker | Right now I know of the Lucent MAX TNT and the Dialogic IMG-1010 |
| 19:01.03 | mocker | Cisco's product line is so big I'm not sure which one fits the bill. ?) |
| 19:01.04 | mocker | :) |
| 19:01.08 | _Corey_ | mocker: Not without searching... If you had asked me about 3 months ago I would have remembered :) |
| 19:01.27 | _Corey_ | I remember one was discontinued though, if that is of any use |
| 19:02.18 | mocker | I may throw the question out to asterisk-users |
| 19:02.19 | ttpears | Qwell: On http://www.voip-info.org/wiki/view/Asterisk+AEL2 they define "Expression Syntax", "Variable Reference Syntax", "Extension Language Syntax", and "AEL". Under "Differences with the original version of AEL" section I can see they explain in part 21 that they now allow "==" to be used in "Expression Syntax", described as: $[...] |
| 19:02.42 | ttpears | But it doesn't mention the "Variable Reference Syntax" :-) |
| 19:03.10 | _Corey_ | mocker: I think one was the successor of the old AS5200 |
| 19:03.12 | ttpears | Which is described elsewhere as "${..}" |
| 19:07.17 | Qwell | Your first problem is taking anything that site says as truth. |
| 19:07.21 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:09.31 | ttpears | lol nice |
| 19:10.04 | ttpears | Well, I'll certainly be buying the definitive guide so hopefully it'll help my facts |
| 19:11.26 | ttpears | Basically, there's a few places that we use "==" in our curly brace syntax, but I'm wondering if that's correct |
| 19:13.50 | *** join/#asterisk roswell (roswell@62.69.14.137) |
| 19:13.54 | leifmadsen | In Asterisk extensions.conf all comparisons are done with a single char, and not a double char, so I would start with that in AEL. |
| 19:16.22 | roswell | hi everyone. would anyone give me a tip why might ${RECORDED_FILE} remain empty after successful completion of Record(/tmp/somefile.gsm,,30,k) though the file itself was created successfully? |
| 19:20.13 | Qwell | roswell: Because you aren't using a dynamic filename |
| 19:21.12 | roswell | hm... sounds reasonably, i'll try that out |
| 19:21.23 | *** join/#asterisk Poincare (~jefffnode@2001:470:d6b3:4::2) |
| 19:23.21 | roswell | Qwell, it worked, thanks |
| 19:24.45 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 19:25.26 | ttpears | Thanks Qwell, I'll let you know how testing goes, and pre-order my copy :-) |
| 19:28.19 | ttpears | Er, thanks leifmadsen and Qwell |
| 19:32.16 | mocker | preorders. |
| 19:41.29 | russellb | \o/ |
| 19:43.41 | *** part/#asterisk ManWithNoName (~ManWithNo@200.242.28.231) |
| 19:46.50 | mocker | leifmadsen: Will the PDF still be online for free? |
| 19:47.17 | mocker | is impatient, book should be here now! :) |
| 19:51.04 | _Corey_ | probably should explore a kindle version or something before the crowds show up with pitchforks in huntsville... |
| 19:51.30 | _Corey_ | ;) |
| 19:52.54 | *** join/#asterisk clintc (~clintc@n128-227-204-35.xlate.ufl.edu) |
| 19:52.57 | *** join/#asterisk jeffgus (~jeffgus@green.zimage.com) |
| 19:54.29 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 20:10.37 | pigpen | Hi all. I have been searching around with nothing but old posts. |
| 20:11.04 | pigpen | Can I have an IAX trunk (or connection, ie: not in trunking mode) pass T.38 ? |
| 20:12.04 | pigpen | my topology: Fax -- Audicodes FXS -----SIP----Asterisk 1.6.2.x-----IAX----Asterisk 1.6.2.x-----PRI |
| 20:12.26 | mocker | emails the list his DS3 question. |
| 20:12.45 | *** part/#asterisk clintc (~clintc@n128-227-204-35.xlate.ufl.edu) |
| 20:13.08 | pigpen | I was running: Fax---Digium(Astiersk)---IAX-----Asterisk---PRI with no issues for a few years now. |
| 20:26.55 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:35.20 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:45.23 | *** part/#asterisk wesphillips (~wphillips@137.237.233.124) |
| 21:02.35 | *** join/#asterisk _zoom_ (~Administr@41.218.57.187) |
| 21:03.26 | _zoom_ | hello, am looking for opensource call center integrated with asteri ? |
| 21:13.47 | *** part/#asterisk _zoom_ (~Administr@41.218.57.187) |
| 21:14.41 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 21:19.17 | *** join/#asterisk Iiiak (~Iiiak@AMontpellier-551-1-9-198.w92-133.abo.wanadoo.fr) |
| 21:35.59 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:10.12 | *** join/#asterisk Pio (~sean@207.181.14.102) |
| 22:10.39 | Pio | can someone recommend a good PPA with asterisk 1.8 for ubuntu 10.10? |
| 22:11.28 | Pio | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages maybe this |
| 22:22.53 | *** join/#asterisk TFGBD (~refhunetg@c-98-225-193-141.hsd1.pa.comcast.net) |
| 22:25.56 | TFGBD | Okay, this might be a stupid question but... Does anyone know of some sort of driver that emulates a voice/fax modem enough to let one dial into an ISP or BBS using only your SIP provider? |
| 22:35.09 | *** join/#asterisk mayeco (~quassel@ubuntu/member/mayeco) |
| 22:57.34 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
| 22:58.27 | ariel_ | evening everyone |
| 23:04.01 | saxa | hopla, sombody knows to tell me if the modem LM-I56N with Motorola chip MDV92XP can work as an FXO ? |
| 23:05.02 | russellb | Pio: yep, that's the official asterisk.org packages for asterisk 1.8 |
| 23:06.03 | Pio | i installed them at they are working great |
| 23:06.04 | Pio | thanks |
| 23:10.18 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 23:17.53 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:27.05 | *** join/#asterisk minaguib (~mina@modemcable098.129-202-24.mc.videotron.ca) |
| 23:45.12 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:50.02 | *** join/#asterisk dlogan (~dlogan@cpc1-nmal18-2-0-cust168.croy.cable.virginmedia.com) |
| 23:50.22 | dlogan | hi |
| 23:50.59 | dlogan | i am trying to get asterisk 1.8 working with ices - streaming mp3 |
| 23:51.35 | dlogan | i have some white noise - currently - via lame |
| 23:52.10 | dlogan | anyone got any pointers - have tried everything on voip-info and also searched forums / digium etc. |
| 23:55.00 | *** join/#asterisk meef (~meef@ip-109-91-17-104.unitymediagroup.de) |
| 23:55.07 | *** part/#asterisk meef (~meef@ip-109-91-17-104.unitymediagroup.de) |
| 23:58.50 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
