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| 01:27.30 | jaskew | Hi all. My old asterisk box is getting a bit elderly (10 yr old deskpro 733Mhz). I think it's time to proactively replace it. I've seen recommendations for the Foxconn R10-D2. Anyone have comments on the quality of that box? Any other boxes that you would recommend? I want something solid. |
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| 01:28.19 | jaskew | I have just a couple VoIP extensions and a VoIP trunk. I have 1 POTS trunk, but don't even use it. |
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| 01:32.10 | jaytee | I just built a * box using an Intel D510MO mainboard and added a low profile Intel NIC to get a second NIC. Board, RAM, 160GB 2.5" sata drive and case cost just under 300 bucks |
| 01:34.58 | *** join/#asterisk guilhermebr (~Guilherme@189.63.57.56) |
| 01:35.41 | jaskew | Thanks jaytee. I suppose HDD and power supply would be the other failure points. Any brand preference on those? |
| 01:45.14 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
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| 01:49.05 | jaytee | I got a Western Digital drive but if you don't need a lot of storage for voicemail then you can get an 40GB SSD drive for around a hundred bucks. I paid 49.99 for this mini-itx case that comes with a 250w PSU |
| 01:52.59 | jaskew | I like that Intel board. No fan... SSD might be good too. I know it's wishful thinking, but I'd like this to last another 10 yrs! |
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| 02:47.04 | maxagaz | hi |
| 02:47.13 | *** join/#asterisk niklas- (~niklas@0173100204.0.fullrate.dk) |
| 02:47.18 | maxagaz | When I make TLS/sRTP calls, it works, but still I get this error message in Asterisk: SRTP unprotect: authentication failure |
| 02:47.20 | maxagaz | Why? |
| 02:48.19 | niklas- | Hey. Shouldn't the result from running 'dundi lookup number@context' return the same results as the DUNDIQUERY() function? |
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| 02:52.36 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 03:35.11 | dke211 | I am seeing some behvior in asterisk i dont understand |
| 03:35.24 | dke211 | maby someone can point me to the right place in the docs.. |
| 03:36.07 | dke211 | when i have an SIP extension dial some extension, the behavior of Dial seems to be different than if i have a DAHDI source do the dialing. |
| 03:36.35 | dke211 | my dial plan is striped down to just about nothing |
| 03:37.31 | dke211 | When dial is called from the DAHDI source, after 4 rings, i get a message to the effec that the mail box is not configured. |
| 03:37.44 | dke211 | rather funny since I dont even have the voice mail object loaded :) |
| 03:37.58 | dke211 | Have I overlooked something simple? |
| 03:39.16 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 03:40.04 | dke211 | any ideas? |
| 03:40.39 | WIMPy | Set verbos to at least 3 and check what's actually going on. |
| 03:41.02 | dke211 | I did that, I even have a telnet session going to the manager port. |
| 03:41.02 | Nugget | telnet is eeeeeeevil! |
| 03:41.07 | dke211 | *laugh* |
| 03:41.40 | dke211 | my project is actually in dealing with external control of asterisk, but I am a bit stumped why a incomming |
| 03:41.53 | dke211 | call only rings the sip extension about 4 times. |
| 03:42.28 | dke211 | from the cli i dumped the actual dial plan, and it was what i put in :) |
| 03:42.44 | WIMPy | So if you see what's going on, you should see the difference. |
| 03:42.55 | dke211 | one would think :) |
| 03:43.46 | dke211 | I'm getting ready to pronounce me as being an idiot, because i must* have overlooked something. |
| 03:44.11 | WIMPy | Then let us have a look. |
| 03:45.14 | dke211 | would you like to have an account on the machine and have at it? |
| 03:45.32 | WIMPy | ~pb |
| 03:45.33 | 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.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. |
| 03:46.04 | dke211 | interisting.... |
| 03:46.08 | WIMPy | Put up your dialplan and the verbose cli output. |
| 03:50.54 | dke211 | here is the dialplan |
| 03:51.47 | dke211 | localhost*CLI> dialplan show |
| 03:52.03 | dke211 | like i said, its striped down to nothing. |
| 03:52.23 | dke211 | the dahdi extension goes to external |
| 03:52.47 | dke211 | The it makes the connection then stops after about 4 rings. |
| 03:53.08 | WIMPy | So where's the output? |
| 03:54.51 | dke211 | ocalhost*CLI> localhost*CLI> |
| 03:54.53 | dke211 | asterisk was run with -vvvc |
| 03:54.54 | dke211 | sorry about the delay there. |
| 03:55.25 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 03:55.28 | WIMPy | is waiting for a link to something to look at. |
| 03:59.51 | dke211 | ahh... try this http://pastebin.com/vGKQpgPM |
| 04:00.07 | dke211 | never used this before, thanks for your patience |
| 04:00.30 | dke211 | better? |
| 04:01.15 | ChannelZ | what's the question again? |
| 04:02.02 | dke211 | when I dial in from a dahdi source, the sip extension will ring 4 times then stop with a message to theeffect that that mail box is not configured! |
| 04:02.24 | dke211 | I disabled the VM system (ie not loading the .so file) |
| 04:02.34 | dke211 | I renamed the directory with the sound files :) |
| 04:02.58 | dke211 | its still* doing it. I figured if it couldnt find the sound files the error message might be helpful. |
| 04:03.12 | dke211 | I must be over looking something - and probably something simple. |
| 04:04.16 | ChannelZ | well the dialplan you posted can't do that so something else is going on. |
| 04:05.16 | dke211 | thats what I am thinking. |
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| 04:05.30 | ChannelZ | what are you calling _from_? |
| 04:06.51 | dke211 | and I also disabled pbx_ael so that is not bringing anthing else in. I am calling from my cell phone to my home extension, that has the wildcard on it. |
| 04:08.12 | dke211 | let me repharse that.. I am calling my home analog line from my cell phone. |
| 04:08.13 | ChannelZ | The message you are hearing isn't coming from Asterisk, based on the dialplan you showed and the console output |
| 04:08.13 | WIMPy | So maybe you have an unconfigured VM from the operator of your landline? |
| 04:08.55 | dke211 | hmmm... intreaguing idea... dont think so, but let me test that out. |
| 04:10.16 | dke211 | oh my! that seems to be the case. |
| 04:10.36 | dke211 | I shut down asterisk and it still* does that. funny thing is that it seems to have started today :) |
| 04:11.01 | dke211 | WIMPy My thanks for pointing out the obvious |
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| 05:04.03 | niklas- | I could use a suggestion for a workaround to this problem. For every SIP user with voicemail defined in voicemail.conf, pbx_config adds a Macro(stdexten..). However i'm using AEL2 which converts 'Set(foo=bar)' to 'Set(LOCAL(foo)=bar)', and LOCAL() cannot be used within a Macro. It doesn't seem possible to make pbx_config use Gosub instead of Macro, at least not without patching asterisk. |
| 05:23.01 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:25.39 | fenrus | lwin 9 |
| 05:57.30 | Corydon76-home | pbx_config has no inherent need to use Macro. You could just as easily be using Gosub there. |
| 05:57.49 | Corydon76-home | It's just that the sample config has a Macro call |
| 05:58.14 | p3nguin | I've never heard of it magically adding a Macro for voicemail, either. |
| 05:58.44 | p3nguin | If you don't want to use a Macro, don't write a Macro. |
| 06:06.00 | niklas- | it's when using users.conf this behaviour happens |
| 06:06.13 | p3nguin | And that's why we don't use users.conf. |
| 06:06.15 | niklas- | my mistake when i wrote voicemail.conf |
| 06:22.05 | *** join/#asterisk recluze (~recluze@119.156.26.103) |
| 06:47.17 | *** join/#asterisk Babak__ (Babak.y@85.133.129.100) |
| 06:47.57 | Babak__ | hi does anyone have localization files for cisco 7941? |
| 06:50.45 | maxagaz | depending whether I use a pem cert or not in my sip software, I get ca-1 as destination port in wireshark for the calls or na-localise, what does those two names mean ? |
| 06:53.46 | maxagaz | does na means not applicable because I'm not using certificate ? |
| 06:54.47 | maxagaz | also, I'm using certificate only on the sip software side, not on the sip phone side, so TLS works, but I don't understand what's hapenning with those certificates |
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| 07:16.47 | nix8n82 | Has anyone had experience with AGI and AMI in c++ and found a good library? |
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| 07:32.00 | recluze | hi, i'm trying to troubleshoot a problem with asterisk. it hangs up after 30 seconds of a call when dialing outside on a sip trunk |
| 07:32.04 | recluze | it works fine on local calls |
| 07:32.35 | recluze | here's the pastebin |
| 07:32.36 | recluze | http://pastebin.com/TeB9xj8T |
| 07:32.44 | fenrus | sounds like a firewall issue |
| 07:32.54 | recluze | i've tried qualify=yes, nat=yes canreinvite=yes |
| 07:33.01 | fenrus | like the firewall eats the packets |
| 07:33.20 | recluze | asterisk is on a live ip |
| 07:33.32 | recluze | and i've tried from quite a few different (client) locations |
| 07:34.22 | recluze | how can I test the packet-eaten theory |
| 07:34.26 | wdoekes2 | chan_sip.c: Received bye, issuing owner hangup |
| 07:34.46 | wdoekes2 | trace the sip packets and see what's going on |
| 07:34.48 | recluze | yes, I noticed that but i'm not sure where that bye is coming from |
| 07:34.54 | recluze | hnn.. |
| 07:35.02 | *** join/#asterisk beebeeep (~miga@85.94.32.98) |
| 07:35.03 | recluze | i'll try that |
| 07:35.13 | wdoekes2 | it might be from the client that did not receive an ack on the 200 or something |
| 07:35.47 | recluze | should I 'sip set debug peer 1000' ? |
| 07:35.52 | recluze | (1000 is my client phone ) |
| 07:35.57 | wdoekes2 | that would work |
| 07:36.02 | recluze | hnn, ok. testing now |
| 07:36.57 | recluze | ok.. call established. waiting for the hangup to come |
| 07:37.52 | recluze | http://pastebin.com/y7ykC2vV |
| 07:37.58 | recluze | the bye is going from my IP |
| 07:38.13 | recluze | 184.72.x.x is my ip |
| 07:38.36 | recluze | but i've removed the timeout settings from advanced network settings |
| 07:38.59 | *** join/#asterisk schmidts (~schmidts@213.235.212.193) |
| 07:39.00 | schmidts | good morning |
| 07:39.01 | ChannelZ | it looks like the BYE is coming from X-lite |
| 07:39.23 | recluze | yeah |
| 07:39.31 | recluze | but the question is, why |
| 07:39.53 | ChannelZ | http://forums.counterpath.com/viewtopic.php?f=1&t=14922 |
| 07:40.30 | ChannelZ | is maybe there some similar wierd timeouts in x-lite RTP? |
| 07:40.41 | ChannelZ | I don't use that thing so.. |
| 07:41.00 | recluze | yes, there is ... but I've disabled that |
| 07:41.35 | ChannelZ | you get audio in both directions when the call is active? |
| 07:41.47 | recluze | yeah. perfect call until it suddenly hangs up |
| 07:42.03 | schmidts | recluze how long after the call started you see the bye? |
| 07:42.05 | recluze | i even tried to keep talking for all the time so that there was actual voice transmission |
| 07:42.11 | recluze | 30 seconds |
| 07:42.19 | ChannelZ | tried a different softphone? |
| 07:42.37 | recluze | no. |
| 07:42.40 | schmidts | T1x64 timeout this happens if for example the 200 OK is not acknowlegded |
| 07:42.40 | recluze | :$ trying now |
| 07:43.09 | schmidts | recluze try a sip debug for this peer and i am sure you will see retransmits of the 200 ok or the ack or something like this |
| 07:43.11 | recluze | meaning asterisk isn't acknowledging the 200 ok to x-lite and x-lite in turn is hanging up after the timeout |
| 07:43.28 | schmidts | yes |
| 07:43.30 | recluze | i did... there were no sip messages while the call was established |
| 07:44.04 | schmidts | recluze ? no sip messages? no invite, no 100 trying 180 ringing or something like this? |
| 07:44.18 | recluze | let me paste the sip debug |
| 07:44.25 | schmidts | pastebin plz ;) |
| 07:44.37 | schmidts | ~pb |
| 07:44.37 | 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.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 07:44.39 | recluze | http://pastebin.com/be4GL87x |
| 07:45.18 | recluze | 164-189 is the call period |
| 07:45.25 | recluze | 192 is the hangup |
| 07:46.16 | schmidts | recluze your nat traversal doesnt work well: |
| 07:46.17 | schmidts | # |
| 07:46.17 | schmidts | Peer audio RTP is at port 10.8.0.6:58106 |
| 07:46.37 | recluze | er... |
| 07:46.44 | recluze | didn |
| 07:46.51 | recluze | didn't understand that |
| 07:47.48 | schmidts | what i can also see is that you have a contact in your invite like this: # |
| 07:47.48 | schmidts | Contact: <sip:2125615702@173.236.222.103> |
| 07:48.13 | recluze | (I do have a vpn in-between... might that be a problem?) |
| 07:49.04 | fenrus | as long as your firewalls are in order and you dont NAT it should not be a problem. |
| 07:49.13 | schmidts | recluze hmm maybe, it looks a little bit strange for me that your sip communication runs over an external IP but your rtp data should run over private ips |
| 07:49.50 | schmidts | you should take a look at the log of xlite itself, maybe you see retransmit there to a different ip like the one your asterisk has in the 10.x.x.x. network |
| 07:49.50 | recluze | configuration problem perhaps? (it's someone else's configs I'm working with...) |
| 07:50.00 | recluze | hnn. |
| 07:50.16 | recluze | i'll try that |
| 07:51.30 | schmidts | you can also try using ngrep on your asterisk server listening for both ips, the external and the internal one but it looks like typical asymetrical routing |
| 07:51.51 | recluze | that went above my head :) |
| 07:52.08 | recluze | i'll try "ngrep" after the x-lite thing though |
| 07:52.30 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 07:53.16 | schmidts | ngrep -t -W byline src host 184.72.232.203 or src host 10.8.0.6 |
| 07:53.21 | schmidts | thats the command you will need ;) |
| 07:54.44 | schmidts | recluze are you using amazon cloud? |
| 07:54.51 | recluze | yeah |
| 07:54.55 | recluze | bad idea? |
| 07:55.30 | schmidts | i have completly blocked the /15 network of amazon EC2 cause most of sip scans originate from this network but if it works for you, why not ;) |
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| 07:58.42 | recluze | well, I was only using the ec2 for the vpn |
| 07:58.48 | recluze | voice ports are blocked in my country |
| 07:58.56 | recluze | the server is somewhere else |
| 08:01.34 | ChannelZ | it occurs to me that IPv6 will make life more complicated for doing things like banning things like Russia for email, where 90% of my spam comes from |
| 08:01.37 | recluze | just got a call working properly without any config change ... :S |
| 08:02.46 | vegbox | Anyone here use Charter for PRI service? |
| 08:02.48 | ChannelZ | although I suppose subnetting works the same anyway. |
| 08:02.50 | vegbox | or XO? |
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| 08:27.41 | maxagaz | how to force the media to go through the asterisk server between two sip phones ? |
| 08:27.58 | maxagaz | (and not only the signaling) |
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| 08:29.52 | ChannelZ | canreinvite=no or directmedia=no depending on what version you are running |
| 08:32.35 | *** join/#asterisk alex84 (~TheUndead@mail2.tikalnetworks.com) |
| 08:34.03 | alex84 | Hi all :) |
| 08:48.06 | alex84 | Quick question: I'm having a problem with registering a SIP trunk between two * boxes, one runnung 1.4 and the other 1.8.... |
| 08:48.34 | alex84 | Can anyone guide me in the right direction to solving this? |
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| 08:59.44 | wdoekes2 | alex84: watch what the console output says.. do the registers come in at all? bad passwords? etc.. |
| 09:00.03 | alex84 | the registers do come in |
| 09:00.20 | alex84 | the problem is that i'm working on the 1.4 side |
| 09:00.40 | alex84 | and i can only communicate with the other admin via emails... |
| 09:00.50 | alex84 | so that makes it a bit mire tricky |
| 09:00.58 | alex84 | *more |
| 09:01.10 | wdoekes2 | do you get response to the registers then? |
| 09:01.47 | wdoekes2 | (sip set debug on) |
| 09:01.48 | alex84 | i get requests from the 1.8, |
| 09:01.59 | alex84 | but they sip debug |
| 09:02.04 | alex84 | *** |
| 09:02.06 | alex84 | sorry |
| 09:04.44 | alex84 | It automatically ignores the INVITEs... |
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| 09:13.45 | wdoekes2 | who ignores which invites? |
| 09:14.09 | wdoekes2 | and is that the first (main) problem you're having? |
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| 09:27.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:47.50 | andylockran | hey guys, I'm writing a perl script to initiate a call between two extensions, but I don't quite 'get' the behaviour I'm seeing. I've managed to successfully originate a call and ring extensionsion 5108, but would like to call a different extension when 5108 picks up. http://dpaste.com/358520/ |
| 09:50.00 | wdoekes2 | andylockran: replace exten with 5105 and add exten => 5105,1,Dial(SIP/5105) to [default] |
| 09:50.47 | wdoekes2 | (oh, and set Priority to 1) |
| 09:51.51 | *** join/#asterisk Tim_Toady (~moi@178.128.134.192.dsl.dyn.forthnet.gr) |
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| 10:02.39 | andylockran | ta |
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| 10:30.41 | *** join/#asterisk evangelion (~manzy_zet@212.183.172.126) |
| 10:30.48 | evangelion | hello |
| 10:31.41 | evangelion | "j" argument in Dial() command forces dialplan to jump to n+101 if called number is busy! |
| 10:31.48 | evangelion | what if i use label? |
| 10:32.28 | evangelion | ie: Dial command is called on position corresponding to label n(dial) |
| 10:36.44 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:36.56 | maxagaz | how to force a peer to register when it takes too much time for it to do it automatically ? |
| 10:40.50 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 10:41.13 | defswork | anyone any experience of shoretel systems ? |
| 10:44.30 | *** join/#asterisk leejohn (~stealth@58.69.36.3) |
| 10:46.13 | leejohn | good day guys, can some one share their experience on deploying asterisk in a WAN scenario does 64 kilobits up/down is really adequate on g.711u codec? |
| 10:46.24 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 10:46.32 | defswork | 64kb ? :o |
| 10:46.49 | leejohn | on a per site |
| 10:47.24 | defswork | how many simultaneous calls you hoping to carry |
| 10:47.48 | leejohn | just one call per site our backhaul is about 6 mb |
| 10:49.06 | leejohn | defswork: i'll rephrase the question, on a 6 mb backhaul how many calls can we squeeze on a g.711u codec |
| 10:50.31 | schmidts | leejohn which backhaul? if you have for example an ATM connection you will have a big overhead but normally with ethernet encapsulation you will have 86kbit/second for each g711u channel |
| 10:51.07 | *** join/#asterisk salimb (~chatzilla@83-244-177-2.cust-83.exponential-e.net) |
| 10:51.58 | tzafrir | evangelion, that option is intended for compatibility with older dialplans |
| 10:52.05 | tzafrir | (pre asterisk 1.2) |
| 10:52.27 | evangelion | tzafrir: what's for 1.8? |
| 10:52.29 | tzafrir | Most likely you don't need it |
| 10:52.42 | schmidts | the normal payload of g711u is 64 kbit + 22 kbit ip header. if you have for example pppoe over an ATM connection you will have 106kbit for each call ;) |
| 10:52.44 | leejohn | schmidts, that's ATM for sure |
| 10:52.54 | tzafrir | If you write a new dialplan for 1.8, you don't need it |
| 10:53.14 | schmidts | leejohn then you have another overhead for ATM too, normaly you can calculate with 100kbit per call |
| 10:54.19 | leejohn | schmidts, that's about 13 kilobytes sustain up/down. |
| 10:54.57 | schmidts | yes |
| 10:55.35 | schmidts | leejohn take a look here: http://pastebin.com/Cj8QSwjS thats a calculation we have done to really know how much bandwith a call would need |
| 10:56.51 | leejohn | schmidts: thank you so much for a wonderful information |
| 10:57.12 | schmidts | but it also depends on how many packets per seconds are used. AFAIK asterisk and also other use 20ms packets so you will have 50 packets per second if you use for example 40 ms packets you will have bigger packets but less overhead overall |
| 10:59.05 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:01.10 | leejohn | schmidts, but if you loose those packets there was a large audio gap right? |
| 11:02.34 | schmidts | yes |
| 11:02.59 | Gugge | i dont know if i would call 1/25 of a second a large audio gap :) |
| 11:03.00 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 11:03.11 | schmidts | btw this calculation is based on g711alaw not ulaw but IMHO this is the same |
| 11:03.30 | schmidts | gugge if you are a fax, then it will be long enough ;) |
| 11:03.42 | Gugge | in my world fax is dead (i wish) :) |
| 11:03.48 | schmidts | :D |
| 11:04.04 | *** part/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 11:04.53 | schmidts | a gap larger than 80ms are recognized by human on a constant audio stream like music, in normal takling you will only hear more than 150ms. if you try it with the 1khz tone you will recognize also 30 or 40 ms |
| 11:07.21 | leejohn | schmidts, so to be safe it's best if we get a 128 kbps / site per call |
| 11:07.55 | Gugge | thats a safe bet yes |
| 11:10.42 | *** join/#asterisk recluze (~recluze@119.156.26.103) |
| 11:10.45 | leejohn | Gugge and schmidts thanks i'll really appreiciate it |
| 11:11.01 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:12.11 | evangelion | tzafrir: sorry, what do you mean? how can I handle busy signals? |
| 11:12.30 | tzafrir | look at DIALSTATUS |
| 11:12.41 | tzafrir | Look for it in the sample dialplan |
| 11:12.45 | evangelion | hum |
| 11:12.53 | evangelion | yes, sounds good! |
| 11:12.55 | evangelion | thank you |
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| 13:38.32 | joachim_- | any1 here familiar with csync2? What does it mean that a file is marked dirty? And is in a dirty state? Getting some annoying error messages :P |
| 13:47.16 | *** join/#asterisk BeeBuu (db81bb1a@gateway/web/freenode/ip.219.129.187.26) |
| 13:47.45 | Dovid | trying to use asterisk to convert over sound files. I get this error: format_wav.c: Unable to set write file size |
| 13:47.56 | BeeBuu | hello,all |
| 13:48.29 | Dovid | is there any other software other than asterisk that can do it ? |
| 13:49.02 | WIMPy | Dovid: sox |
| 13:50.54 | Dovid | WIMPy: SOX will do G729 ? |
| 13:51.42 | WIMPy | Probably not. |
| 13:51.47 | Dovid | :( |
| 13:51.57 | Dovid | thats what i want to do. i want to cut down on transcoding |
| 13:52.25 | WIMPy | I guess you need to use Asterisk for that. |
| 13:52.35 | BeeBuu | i wrote this exten when someone call in: exten=1111,1,dial(sip/1000&sip/1008,,TtG(a^b^1)),how can i know which sip user pickup the call? |
| 13:53.41 | *** join/#asterisk pecenipicek (~pecenipic@cpe-109-60-73-149.zg3.cable.xnet.hr) |
| 13:53.47 | pecenipicek | good day gentlemen. |
| 13:54.09 | pecenipicek | Is there a way to get an ip address of a specific interface on a machine from inside asterisk? |
| 13:54.32 | WIMPy | pecenipicek: No |
| 13:54.50 | pecenipicek | Okay. |
| 13:55.20 | WIMPy | Asterisk does not care about interfaces at all. |
| 13:55.38 | *** join/#asterisk BeeBuu_ (db81bb1a@gateway/web/freenode/ip.219.129.187.26) |
| 13:55.38 | pecenipicek | its however possible to get asterisk to call a specific script and get its output afterwards, right? |
| 13:55.48 | BeeBuu_ | anyone help me please? |
| 13:55.51 | pecenipicek | Asterisk maybe doesnt care, but i do :p |
| 13:56.08 | WIMPy | pecenipicek: Sure |
| 13:56.42 | pecenipicek | WIMPy, kk. |
| 13:57.02 | BeeBuu_ | how can know who is pickup when dial(sip/1000&sip/1001) ? |
| 13:57.04 | pecenipicek | this is still a continuation of my whole idiocy with VPN stuff |
| 13:57.38 | *** join/#asterisk kaii (~kh@mail.ciphron.de) |
| 13:57.50 | joachim_- | any1 here familiar with csync2? What does it mean that a file is marked dirty? And is in a dirty state? Getting some annoying error messages :P |
| 13:58.18 | WIMPy | pecenipicek: Maybe you should put another tunnel upon the other tunnel to get a static IP. *cough* |
| 13:58.25 | pecenipicek | :p |
| 13:58.37 | kaii | is working on a project soft migrating all phones from a MS OCS 2007 setup over to asterisk 1.8 .. <3 |
| 13:59.43 | pecenipicek | well, the only thing thats left to fix up the calls from the remote box is to get it to put the proper ip address from the tun0 interface when performing a dundi lookup. |
| 13:59.53 | pecenipicek | and not the thrice damned local ip. |
| 14:00.49 | WIMPy | That sounds interesting... |
| 14:01.10 | pecenipicek | its a very bizzare and particular setup. |
| 14:01.30 | pecenipicek | i'm actually amazed that noone ever did it, but oh well |
| 14:02.51 | WIMPy | If someone had the same issue on the internet, they just use dyndns, I guess. |
| 14:02.59 | pecenipicek | yeah |
| 14:03.18 | pecenipicek | and corporate-style deployments arent exactly known for sharing their discoveries. |
| 14:03.31 | devdvd | hi all, i am using asterisk 1.6.2.15, have my queue members in a mysql database, i want to use a linear call queue but the problem seems to be that asterisk wants to order the queue members via interface (so say i want to have the order be SIP/802 > SIP/800 > SIP/801, asterisk will actually order it as 800 > 801 > 802) so my queue gets all messed up |
| 14:03.40 | devdvd | any thoughts on how to make asterisk query the queue members off a different field (like uniqueid or even a custom field) |
| 14:04.34 | pecenipicek | Most of my DUNDi and IAX stuff is taken from this document, which has proven time and time again its awesomeness. http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/ |
| 14:04.49 | Katty | gooooooooooooood morning all you wonderful people |
| 14:04.50 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 14:04.54 | Katty | it's THURSDAY! |
| 14:04.59 | Katty | that means tomorrows FRIDAY! |
| 14:05.20 | devdvd | yes it is! I I have a half day of work today and off tomorrow :) |
| 14:05.32 | WIMPy | pecenipicek: I think you could do some trickery by supplying a peer name instead of a hostname via dundi. |
| 14:06.32 | pecenipicek | i'm supplying an IP address, since hostnames dont traverse the vpn from my end. |
| 14:06.52 | pecenipicek | hrrm... |
| 14:06.57 | pecenipicek | i might test it |
| 14:06.57 | leifmadsen | The Asterisk Cookbook has finally been started. |
| 14:07.08 | leifmadsen | Katty: morning! |
| 14:07.11 | WIMPy | pecenipicek: Yes, then use a dynamic peer perhaps? |
| 14:07.14 | Katty | good morning mister leifmadsen! |
| 14:07.17 | Katty | hugs on leifmadsen |
| 14:07.23 | Katty | russellb: be safe on your trip!! |
| 14:07.42 | russellb | Katty, thanks :-) |
| 14:07.44 | leifmadsen | Katty: thank you! |
| 14:08.31 | pecenipicek | WIMPy , you lost me. |
| 14:08.47 | pecenipicek | lemme describe the network build up at the moment. |
| 14:08.59 | pecenipicek | there are two servers locally, with fixed ip addresses. |
| 14:09.14 | pecenipicek | theres a single remote server, also with a fixed address. |
| 14:09.54 | *** join/#asterisk b0ot (~Jinxed---@147.177.57.8) |
| 14:10.20 | pecenipicek | there are 2 vpn tunnels to the remote networks, which for the reasons of network admin incompetency(not mine and sadly not fixable), get IP addresses which change every 90 minutes or with every new connect via the vpn |
| 14:10.44 | b0ot | Could I put asterisk on an old computer and set it up to work with google voice with an analog phone and a ATA to get free calling at my apartment? |
| 14:10.49 | BeeBuu_ | how can know who is pickup when dial(sip/1000&sip/1001) ? |
| 14:11.43 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 14:11.57 | pecenipicek | the general idea is to have dundi lookups traverse all the segments around properly. local calls work perfectly, at both ends. inter-server calls work in only one way. (from us to the company's remote server), but not the other way around. |
| 14:12.49 | pecenipicek | the reason they dont work from the other way is because of using the "priv => dundiextens,0,IAX2,iaxuser1:${SECRET}@192.168.2.102/${NUMBER},nopartial" for dundi lookups from one of the boxes. |
| 14:13.36 | pecenipicek | now, when the remote box initiates a dundi lookup, it gets the response with the IP address of the machine thats in the local network. |
| 14:13.43 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 14:14.59 | pecenipicek | what i plan to do is simply replace the hardcoded address defined in the mapping with a call to the script which outputs the IP address that the tun0 interface on the local box has and it puts that ip address instead of the local one. |
| 14:15.05 | kaii | anyone knows a plausible reason why DIALSTATUS returns "CONGESTION" when a sip user returns 480 "Do Not Disturb"? shouldnt that be "BUSY"? |
| 14:15.09 | leifmadsen | b0ot: sure you could! :) |
| 14:15.30 | leifmadsen | kaii: could just be the way it is coded, in which case a bug being filed may be appropriate |
| 14:15.56 | WIMPy | pecenipicek: Soory, was AFK. Yes. My idea was to supply a peer name via dundi that is configured as dynamic and you register aginst it. |
| 14:16.06 | pecenipicek | and the script in question is this: /sbin/ifconfig tun0 | grep 'inet addr:' | cut -d: -f2 | awk '{ print $1}' |
| 14:16.12 | leifmadsen | b0ot: wouldn't really require a lot of horsepower... a P3-800 with like 512MB of RAM would be lots of power for that system. Just a bit slow to compile, but that's no biggie. |
| 14:16.14 | pecenipicek | which does exactly what i need |
| 14:16.29 | kaii | leifmadsen: of course it is the way it is coded, that is why i was asking for plausible reasons before filing a bug. this behaviour seems to be consistent in all asterisk version, but looks wrong to me. |
| 14:16.37 | leifmadsen | runs off to work on his Asterisk presentation for this afternoon. |
| 14:17.00 | leifmadsen | kaii: I can't think of any particular reason why it would return CONGESTION |
| 14:17.00 | pecenipicek | hrrrmmm. |
| 14:17.40 | pecenipicek | isnt congestion used as a catch-all when asterisk runs out of other intelligent responses? |
| 14:18.04 | kaii | pecenipicek: yes |
| 14:18.27 | pecenipicek | kaii: this might help a bit... http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 14:19.13 | leifmadsen | kaii: just file an issue with a SIP trace and the dialplan in use, along with the output of DIALSTATUS and we'll get it triaged and either fix or closed |
| 14:19.30 | leifmadsen | if closed, at least you'll know why CONGESTION is right instead of BUSY |
| 14:19.32 | pecenipicek | WIMPy, could you maybe give a quick mockup of how that'd work? |
| 14:20.32 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 14:23.02 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:10c8:5c90:d53:7f5d) |
| 14:23.41 | kaii | leifmadsen: will do, thx |
| 14:24.22 | beek | good day, everyone |
| 14:25.06 | Dovid | file.c: Unable to open /root/greeting.wav |
| 14:25.37 | Dovid | why would I get this when trying to convert a file ? could it be a permissions issue ? |
| 14:25.49 | beek | Yes, if Asterisk is not running as root. |
| 14:26.34 | Dovid | beek: Why would that be an issue if asterisk is running under the same user of the files permission ? |
| 14:26.41 | Dovid | beek: Or is it just how asterisk works ? |
| 14:26.51 | beek | Dovid: No, it's how *nix works. |
| 14:27.10 | beek | Just because the user owns the file, /root is /root's home directory and there would be no perms there. |
| 14:27.36 | kaii | move it to /tmp and you'll be fine |
| 14:28.09 | Dovid | ok. didnt think of that. let me move to to tmp and test |
| 14:28.18 | beek | '/x/y/z' if user has perms to z but not y or x then there's no access to z |
| 14:30.04 | pecenipicek | now, the bigger issue is, how do i actually run a external script from asterisk... |
| 14:30.19 | pecenipicek | goes off to check the asterisk functions page on voip info |
| 14:31.24 | Tozz_ | AGI() |
| 14:33.28 | leifmadsen | or SHELL() or System() :) |
| 14:33.57 | leifmadsen | oh darn no russellb |
| 14:34.10 | leifmadsen | Asterisk Cookbook Info: http://leifmadsen.wordpress.com/2011/01/27/asterisk-cookbook-is-on-we-think/ |
| 14:35.53 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f16f:925a:9ed3:421a) |
| 14:35.56 | pecenipicek | leifmadsen , thank you, that helps |
| 14:36.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:36.36 | Dovid | kaii: That worked :) how ever when trying to convert the dir where i store my sound files it does not work. if i move it to /tmp then it works |
| 14:39.16 | Dovid | kaii: Hmm. Now I get format_wav.c:224 update_header: Unable to set write file size |
| 14:39.54 | kaii | Dovid: read what beek said and read up on unix/linux file system permissions then .. |
| 14:40.23 | Dovid | kaii: i am doing it in /tmp |
| 14:41.29 | Dovid | beek: I did not understand "'/x/y/z' if user has perms to z but not y or x then there's no access to z" |
| 14:41.39 | Dovid | ah. |
| 14:41.49 | Dovid | now i am not getting a perms error |
| 14:42.55 | beek | Dovid: 'z' is a file in the directory '/x/y' Unless you have permissions to '/x' and '/x/y' then what perms are assigned to 'z' are irrelavant. |
| 14:43.25 | Dovid | beek: Thank you very much. i should have perms but I will look in to it |
| 14:43.36 | beek | Dovid: http://www.linux.com/learn/tutorials/309527-understanding-linux-file-permissions |
| 14:44.02 | Dovid | . /tmp should be writeable by ev1. correct ? |
| 14:44.06 | beek | yes |
| 14:44.27 | beek | If there has been no tampering to the directory permissions, that is. |
| 14:45.12 | Dovid | beek: so if Asterisk has permission to write to z (/x/y/z) and records files there, to convert I need access to /x/\y/z ? |
| 14:45.13 | beek | I'd take the moment and read the contents of the link I just sent to you. One page -- explains it well. |
| 14:45.17 | Dovid | ok |
| 14:45.27 | Dovid | or i can just run asterisk as root ;) |
| 14:45.39 | beek | slaps Dovid with a trout |
| 14:47.36 | pecenipicek | is running 2 servers as root and one as an ordinary user... |
| 14:47.38 | Tozz_ | mmm.. trout! yum |
| 14:47.58 | beek | slaps pecenipicek with a trout |
| 14:48.13 | pecenipicek | slaps beek with crappy network admins |
| 14:53.15 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:53.15 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:56.44 | saxa | hi, can anybody advise why fail during make install here ? |
| 14:56.47 | saxa | http://pastebin.com/8yF94SjQ |
| 14:57.06 | saxa | It seems its trying to create a sample voice message |
| 14:57.13 | saxa | i'm not sure about it |
| 14:57.35 | Dovid | hmph. I don't think I have permissions issues since the file has been created. the issue is that when i do to g729 the file is empty |
| 14:57.43 | Dovid | could it be the version of asterisk ? |
| 14:57.50 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-irnvfdttwqtshydw) |
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| 14:58.18 | saxa | ok, its not make install its during make samples. |
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| 15:00.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:00.43 | Dovid | ok. found the issue. |
| 15:01.01 | Dovid | i need to su to user of file then do asterisk -r then convert file |
| 15:08.13 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:11.36 | evangelion | what's MASTER_CHANNEL? Is it the channel of the incoming side during a SIP-UA<-asterisk->SIP-UA call? |
| 15:17.40 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 15:18.12 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 15:20.51 | krion | hi guys |
| 15:21.10 | krion | do you know why i cant execute g729 show licenses using ami |
| 15:21.21 | krion | even with all read priv |
| 15:21.54 | *** join/#asterisk MrSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 15:24.55 | evangelion | anyone knows the difference between master and slave channel? |
| 15:25.29 | evangelion | i can't understand how HASH(SIP_CAUSE,<slave-channel-name>) is intende to be used |
| 15:26.38 | Katty | bonks Qwell |
| 15:26.42 | Katty | Qwell: i /am/ working! |
| 15:31.24 | *** join/#asterisk MrSmurf (~MrSmurf@unaffiliated/mrsmurf) |
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| 15:36.37 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 15:42.08 | evangelion | ok this seems to work |
| 15:42.11 | evangelion | ${HASH(SIP_CAUSE,${SHARED(outChannel,${CHANNEL})})} |
| 15:43.13 | evangelion | ${SHARED(outChannel,${CHANNEL} is set in "call connect macro" |
| 15:43.25 | evangelion | and it reflects the name of the outgoing channel |
| 15:44.18 | evangelion | but it's pretty useless since in this way you can access the outChannel name only if call has been answered |
| 15:44.50 | evangelion | is it possible to get the name of the outgoing channel even if call han not been answered? |
| 15:45.06 | *** join/#asterisk umay (~chris@67-41-151-157.hlrn.qwest.net) |
| 15:45.29 | WIMPy | evangelion: What are you trying to do? |
| 15:46.43 | evangelion | WIMPy: "Asterisk 1.8 will allow to read SIP response codes in the dialplan via ${HASH(SIP_CAUSE,<slave-channel-name>)}" |
| 15:47.13 | evangelion | and indeed it works i can see "SIP 200 OK" for answered call |
| 15:47.33 | evangelion | but what for non-answered? does outgoing channel is created even if Dial() command fail? |
| 15:48.15 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:91d4:fec0:6eb4:6717) |
| 15:48.32 | WIMPy | You want to get more than DIALSTATUS and HANGUPCAUSE, i.e. the text? |
| 15:49.21 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:49.30 | evangelion | WIMPy: http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 15:50.24 | WIMPy | What about that? |
| 15:51.01 | evangelion | "Asterisk 1.8 will allow to read SIP response codes in the dialplan via ${HASH(SIP_CAUSE,<channel-name>)}. Additionally make sure you're using the destination channel, not the source channel. " |
| 15:52.53 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
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| 16:02.26 | jpcansa | does anyone knows how can i record/monitor calls on demand? i need to Q&A my agent´s conversations |
| 16:04.51 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 16:08.18 | Dovid | jpcansa: CHanspy |
| 16:10.23 | ariel_ | ping tzafrir, how can I get in contact with someone for support over 2 XR3000 and TwinStar setup? |
| 16:12.23 | jpcansa | Dovid; thanks i´ll take a look at it |
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| 16:32.31 | webar7 | wtf gotomeeting.com stole asterisk's icons |
| 16:32.41 | webar7 | bastards |
| 16:32.49 | *** part/#asterisk webar7 (~webart@CPE0080c8f208a5-CM001371173cf8.cpe.net.cable.rogers.com) |
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| 16:52.36 | jamko | Hello. When using asterisk with sip and realtime, what is the absolute minimum that must still remain in the sip.conf flat file? It look like all it needs is a [general] context, but I want to be sure. |
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| 17:00.18 | *** join/#asterisk powerunits (~aa@119.152.43.230) |
| 17:00.29 | powerunits | hello ppls... |
| 17:00.59 | powerunits | plz could any one tell me in 1 MB how many calls i can make at same time using GSM codec? |
| 17:01.03 | powerunits | plz guide me thanks |
| 17:04.24 | andylockran | guys, running FreePBX 2.8 and trying to setup followme. Doesn't appear to be working at all.. any ideas? |
| 17:04.34 | Qwell | andylockran: #freepbx |
| 17:04.57 | jamko | powerunits: http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 17:06.02 | powerunits | jamko: thanks let check that |
| 17:06.42 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.12.182) |
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| 17:17.11 | Qwell | <webar7> wtf gotomeeting.com stole asterisk's icons |
| 17:17.14 | Qwell | Asterisk has icons? O.o |
| 17:18.28 | WIMPy | I guess that was an ad. |
| 17:21.55 | cusco | hello |
| 17:22.01 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 17:22.02 | cusco | simple questiona bout unique ID |
| 17:22.33 | cusco | I noticed that a channel that derivates from another channel maintains the first part (before the dot .) and only changes after the . |
| 17:22.40 | cusco | (using 1.6) |
| 17:23.28 | cusco | can I assume that every call has a different first part, unless it is a derivated channel?? |
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| 17:36.16 | p3nguin | There aren't any "icons" on gotomeeting.com anyway. I checked. |
| 17:36.43 | WIMPy | So it's the same as the Asterisk icons. |
| 17:36.51 | p3nguin | Yes, none. |
| 17:40.44 | Qwell | how...meta |
| 17:42.19 | Katty | meta you in a minute. |
| 17:42.35 | Qwell | Katty: You know where the key is |
| 17:42.40 | *** join/#asterisk garymc (~chatzilla@host81-148-2-229.in-addr.btopenworld.com) |
| 17:43.21 | Qwell | Katty: also, I totally didn't IM you this morning and leave right afterwards |
| 17:43.27 | Qwell | definitely not |
| 17:51.38 | Katty | :P |
| 17:54.54 | *** join/#asterisk joel_oliveira (~chatzilla@alpes.nortenet.pt) |
| 17:58.05 | pabelanger | ~book |
| 17:58.05 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
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| 18:46.30 | Kobaz | so, i have a polycom 650 and a 670, and they both exhibit some weird 'static' type noise when talking with an ouside caller (sip-t1 gateway) |
| 18:46.47 | Kobaz | we have 50 other phones (331's) on the same box, that don't have this issue |
| 18:47.14 | Kobaz | when the person on the 650 is talking, there's like a sssssh sound for about 1 second that plays back in the handset |
| 18:47.47 | Tozz_ | perhaps different codec? |
| 18:48.03 | Tozz_ | or comfort noise setting? |
| 18:48.10 | Kobaz | hmm, maybe comfort noise |
| 18:48.47 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:49.22 | Kobaz | i doubt it's codec |
| 18:49.25 | Kobaz | everything is forced to ulaw |
| 18:55.48 | *** join/#asterisk azertyu (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 18:55.51 | azertyu | hello there |
| 18:56.22 | azertyu | i installed asterisk on ubuntu server |
| 18:56.59 | azertyu | i create an extension which is 111 when i try to dial the extension |
| 18:57.55 | azertyu | i got this : <SIP/111-00000003> Playing 'prepaid-enter-pin-number.gsm' (language 'en') on my softphone |
| 18:58.10 | azertyu | i got this : <SIP/111-00000003> Playing 'prepaid-enter-pin-number.gsm' (language 'en') on my asterisk cli |
| 18:58.56 | azertyu | but on the softphone there is no voice is that normal, i mean i don't hear the voice asking me to " enter prepaid pin number " |
| 18:59.07 | azertyu | what could be wrong on my installatioN ? |
| 18:59.25 | p3nguin | If you created extension 111, then you obviously told extension 111 to play that sound file. |
| 18:59.49 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:00.07 | azertyu | yes but i can't hear the voice |
| 19:00.29 | azertyu | i tested with softphone & hardphone, i can't hear the voice |
| 19:00.51 | p3nguin | Is the phone in the same network as Asterisk? |
| 19:01.06 | azertyu | no |
| 19:01.22 | p3nguin | Is asterisk behind a NAT and/or a firewall? |
| 19:01.37 | azertyu | no there is no firefall on the server |
| 19:01.51 | p3nguin | Asterisk connects directly to the internet? |
| 19:01.59 | azertyu | correct |
| 19:02.57 | p3nguin | Pastebin the output of "iptables -L -v" from the server. |
| 19:03.10 | azertyu | just a sec |
| 19:03.16 | *** join/#asterisk nix8n82-phone (~AndChat@63.162.28.112) |
| 19:05.30 | azertyu | here is the output of iptable http://paste.ubuntu.com/559154/ |
| 19:05.54 | p3nguin | Okay, that's good. |
| 19:06.35 | p3nguin | Is the phone you're using behind a NAT? |
| 19:06.42 | azertyu | yes correct |
| 19:06.53 | azertyu | whY ? |
| 19:07.02 | p3nguin | Did you correctly configure asterisk to talk to your phone behind NAT? |
| 19:07.29 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 19:07.34 | p3nguin | Not having audio is most commonly a NAT problem. |
| 19:08.30 | azertyu | well, just note that i allready make some call with same configuration |
| 19:08.39 | p3nguin | Did you have audio? |
| 19:08.55 | azertyu | yes i can call all of my extension |
| 19:09.04 | azertyu | and it work welll |
| 19:09.11 | p3nguin | Then the question is: what did you change? |
| 19:09.53 | azertyu | the problem is when i setup the voucher redeem functionality on my asterisk |
| 19:10.37 | azertyu | when i try to make a call i got this line : <SIP/111-00000003> Playing 'prepaid-enter-pin-number.gsm' (language 'en') on my asterisk cli |
| 19:10.56 | azertyu | but on my softphone it don't ask me to enter the pin code |
| 19:11.21 | azertyu | for me the .gsm file not working correctly |
| 19:13.13 | azertyu | <PROTECTED> |
| 19:13.35 | azertyu | or i don't know what "s really wrong ? |
| 19:14.05 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 19:15.13 | BlackBishop | how would I go if I'd want the user who calls in an extension to enter some numbers followed by # and how would I get and use those numbers ? |
| 19:16.04 | _Corey_ | BlackBishop: "core show application Read" |
| 19:16.14 | BlackBishop | thanks :) |
| 19:16.18 | Tozz_ | google for "asterisk waitexten" |
| 19:18.11 | BlackBishop | yup, read was the one ! :) |
| 19:18.15 | azertyu | how to check .gsm file working correctly ? |
| 19:21.11 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:21.20 | p3nguin | azertyu: I use the Playback() application in dialplan to check sound files. |
| 19:22.07 | azertyu | i don't know how to do |
| 19:22.40 | p3nguin | exten => 5555,1,Playback(some-sound-file) |
| 19:22.54 | azertyu | i paid for setup & service to star2billing but that man not |
| 19:23.04 | azertyu | configure it correctly |
| 19:23.19 | p3nguin | I really don't like billing software. It's almost all problematic. |
| 19:23.20 | azertyu | he asks me for more money |
| 19:23.58 | _Corey_ | We wrote our own billing gateway about 6 years ago |
| 19:24.31 | _Corey_ | I think a lot of the packages out there try to accommodate too many scenarios |
| 19:24.49 | p3nguin | My problem is that they mangle the asterisk config files. |
| 19:25.24 | _Corey_ | Hm, yeah, I use an AGI |
| 19:26.01 | _Corey_ | one line in the dial plan : exten => _X.,1,AGI(xxx) |
| 19:26.04 | _Corey_ | :) |
| 19:26.25 | p3nguin | That's certainly not bad. |
| 19:26.31 | azertyu | well you should know the problem for my case ? |
| 19:26.52 | _Corey_ | azertyu: sadly, probably not. Sorry |
| 19:26.56 | azertyu | i got this line on my cli : <SIP/111-00000003> Playing 'prepaid-enter-pin-number.gsm' (language 'en') |
| 19:27.12 | azertyu | but no voice asking me to enter the pin code |
| 19:27.18 | azertyu | what's wrong ? |
| 19:27.37 | _Corey_ | I think p3nguin had suggested your audio may be getting lost due to NAT |
| 19:27.46 | _Corey_ | That's most likely what's happening |
| 19:28.09 | _Corey_ | Make sure you can do basic things before trying to debug a billing system |
| 19:28.16 | azertyu | so what are the prot i have to open ? |
| 19:28.22 | p3nguin | Unless I see the sip.conf and verify it is configured for NAT, I will continue to assume that is the problem. |
| 19:28.25 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 19:29.00 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 19:29.13 | _Corey_ | yeah, I would google "asterisk sip nat" and explore what you find |
| 19:29.25 | p3nguin | ~sipnat |
| 19:29.25 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 19:29.34 | _Corey_ | even better :) |
| 19:30.16 | *** part/#asterisk salimb (~chatzilla@83-244-177-2.cust-83.exponential-e.net) |
| 19:30.49 | p3nguin | Of course that varies slightly for asterisks which are directly connected and the phones are behind NATs elsewhere. |
| 19:31.39 | citywok | every once in a while random calls will drop out of our conference bridge. Auto fallthrough, channel 'SIP/1504-000014bc' status is 'UNKNOWN' |
| 19:32.09 | citywok | any ideas? is it time to just quit using meetme? lol |
| 19:32.34 | citywok | 4 people were in this one, 2 dropped. 1 internal, 1 external. we had 2 internal and 2 external on so it's not likely a network issue. |
| 19:32.38 | _Corey_ | citywok: Is there a DTMF exit option enabled? |
| 19:32.43 | p3nguin | I've never had a problem with MeetMe, but I also have low call volume. |
| 19:33.01 | _Corey_ | maybe some random tone is making its way in |
| 19:33.28 | p3nguin | I'd try watching the console for clues on what is going on. |
| 19:33.44 | azertyu | i got freepbx sip.conf can i post that ? |
| 19:34.24 | p3nguin | There aren't many people here that care about FreePBX configuration. |
| 19:34.39 | WIMPy | ... or understand them. |
| 19:34.40 | p3nguin | There's an entire channel dedicated to FreePBX just for things of that nature. |
| 19:34.56 | citywok | _Corey_: nobody has the conference room dialpad nearby to even bump it |
| 19:35.13 | citywok | the remote client i can't speak for |
| 19:35.49 | _Corey_ | citywok: Should be easy enough to rule out with some extra debugging |
| 19:36.12 | p3nguin | s/extra/very\ basic/ |
| 19:36.27 | _Corey_ | I had some strange stuff happen like this on a really overloaded system once where audio was getting jumbled |
| 19:36.48 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 19:37.15 | bmoraca_work | getting these errors trying to do t.38 passthrough: "Cannot calculate far_max_ifp before far_max_datagram has been set". asterisk is in the media path. however, i'm not able to manually specify max datagram size because that value is "no longer supported". could a version upgrade potentially solve the issue? i'm on an old 1.6.2 version |
| 19:37.25 | _Corey_ | Meetme is pretty solid by itself |
| 19:38.09 | imox1234 | I want to install asterisk mysql addon on centos but I need a package to install that but I forgot whiche someone has any idea? |
| 19:39.15 | imox1234 | libxml2-devel |
| 19:41.04 | imox1234 | mysql-devel |
| 19:41.05 | imox1234 | :D |
| 19:41.07 | imox1234 | thx |
| 19:42.42 | citywok | _Corey_: I'm now recording all of my conference calls so i can go listen to them after they drop and see what happened |
| 19:43.03 | citywok | it seems to happen once or twice a month, but we have 5-20 conference calls a day |
| 19:43.19 | _Corey_ | citywok: Keep in mind DTMF will likely be inaudible in the recording |
| 19:43.43 | citywok | I record all our agent calls and i can hear the DTMF tones being pressed, although it's impossible to tell WHAT tone. lol |
| 19:43.56 | p3nguin | I have at least one MeetMe conf every week, and I have never experienced that behavior. |
| 19:44.15 | citywok | most of ours are only 3-5 people but there is the occasional 20-30 person conf |
| 19:44.37 | _Corey_ | Just a theory on the DTMF, I think the best bet would be to turn on a good amount of debugging and then grab a thread trace from the full log for an affected call |
| 19:44.44 | citywok | I just need to upgrade our prod server to 1.8 with the new confbridge module out of SVN. lol |
| 19:45.11 | p3nguin | uses 1.4 and MeetMe |
| 19:45.24 | citywok | _Corey_: that's problematic b/c a debug log spews massive amounts of data on the production server, and with it only happening once every few weeks it'd be insane. lol |
| 19:45.44 | _Corey_ | shurgs |
| 19:45.54 | _Corey_ | shrugs too |
| 19:45.59 | _Corey_ | :) |
| 19:46.25 | citywok | the debug log @ 3 is like a megabyte a second. lol |
| 19:47.37 | citywok | it spews millions of these: audiohook.c: Write factory 0x7fe458194a88 was pretty quick last time, waiting for them. |
| 19:48.00 | Katty | hellllllllllloooooooooooooo nurse!!! |
| 19:48.08 | _Corey_ | Hmmm, maybe the core debug could be turned down |
| 19:48.12 | sbrath | I'm having occasional echo on a DAHDI->SIP call where the caller calls person-A ( with echo ) and then person A transfers the call to person B ( and they have no echo ). This seems to tell me that the DAHDI channel is clean. |
| 19:48.31 | citywok | it's only set to 3 right now |
| 19:48.40 | citywok | v&d are @ 3 |
| 19:48.44 | _Corey_ | Yeah, I think you'll get enough with it set to 0 actually |
| 19:48.55 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 19:49.06 | _Corey_ | Just make sure DTMF debugging is going to the same log |
| 19:51.34 | p3nguin | sbrath: Are you hearing the echo on the POTS device or the SIP device? |
| 19:54.43 | citywok | i wish you could log sip directly to its own log file. man that would be nice. |
| 19:54.48 | citywok | sip debug that is |
| 19:55.11 | _Corey_ | I just use ngrep for that |
| 19:55.19 | p3nguin | I guess you could do it without much trouble. |
| 19:55.58 | WIMPy | would prefer a verbosity level for sip debug. |
| 19:56.00 | p3nguin | Use `script` or something similar; run asterisk -r; and then turn on the sip debug. |
| 19:56.26 | _Corey_ | Nobody's used 'ngrep' ? |
| 19:56.47 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 19:56.47 | citywok | i use grep. what's ngrep? lol |
| 19:56.57 | p3nguin | Describe your process... maybe it is better than my idea. |
| 19:57.22 | _Corey_ | lol, ok... ngrep can be had at http://ngrep.sourceforge.net/ |
| 19:57.46 | _Corey_ | so: ngrep host x.x.x.x and port 5060 -W byline -t > file |
| 19:58.04 | _Corey_ | if x.x.x.x is a particular phone you're troubleshooting |
| 19:58.08 | p3nguin | pacman -S ngrep gets me ngrep. :) |
| 19:58.16 | citywok | ah interesting. yea i don't have a single phone issue so it's just all of them. lol |
| 19:58.25 | citywok | i suppose i could just use tcpdump and do the same thing too |
| 19:58.27 | _Corey_ | otherwise just do: ngrep port 5060 -W byline -t > file |
| 19:58.51 | _Corey_ | Yeah, it's a little easier than tcpdump by itself |
| 19:58.57 | _Corey_ | it uses the same libs |
| 19:59.31 | citywok | gotcha |
| 20:00.42 | citywok | nice, it gives decent output too. i'll just use this temporarily and remember to reset it daily. lol. |
| 20:01.05 | p3nguin | crontab |
| 20:01.15 | _Corey_ | It's helpful in trying to find random crazy stuff |
| 20:01.26 | sbrath | p3nguin: on the SIP device ( according to the user ), I'm preparing to start recording channels, if I record the channel with dahdi_ cmds, and I head no echo... what will that mean? |
| 20:01.55 | p3nguin | sbrath: If the SIP device hears the echo, try turning the volume down on that phone. |
| 20:02.14 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 20:02.30 | sbrath | they say that when it happens it sounds like you can hear yourself talking in echo, but the remote side hears nothing. |
| 20:03.04 | p3nguin | SIP doesn't echo, and you've eliminated the Dahdi channel as having echo... so it has to be something in the phone. |
| 20:03.22 | sbrath | it's a Yealink T28 |
| 20:03.38 | p3nguin | Usually, turning down the volume helps. |
| 20:03.51 | sbrath | The volume on the ear-piece? |
| 20:03.54 | p3nguin | yes |
| 20:04.08 | sbrath | Can Jitter have anything to do with this? |
| 20:04.09 | p3nguin | phone/speaker, whatever has the echo |
| 20:04.46 | p3nguin | I doubt jitter will create an echo. |
| 20:04.55 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 20:04.57 | sbrath | I've noticed that in one instance, I had the person run a new home run of cat 5e from the switch to the phone and then the problem went away. |
| 20:05.29 | sbrath | I've seen some of the 10 year old Cat-5 was ran in the same tubes as the AC ( EEEEK ) |
| 20:05.43 | p3nguin | I guess they didn't know it was a bad idea. |
| 20:05.47 | p3nguin | Or they didn't care. |
| 20:06.03 | sbrath | The installer they hired to do it 10 years ago was a virgin to the wiring game... |
| 20:06.17 | p3nguin | Ah, so a combination of the two. |
| 20:06.35 | sbrath | And it's all burried in Concret in Steel tubes..... |
| 20:06.44 | p3nguin | They didn't know they needed to care because it is a bad idea. |
| 20:06.48 | sbrath | I'm asking them to re-pull over the drop ceiling. |
| 20:07.36 | sbrath | can the noise/cross-talk on the ethernet caus the phone to loose frames and cause the issue? |
| 20:08.37 | sbrath | P.S. I'm looking for asterisk people in wisconsin to help on projects, if anyone knows anyone have them ping me... |
| 20:09.03 | p3nguin | Why would I need to be in Wisconsin to work on Asterisk? |
| 20:09.03 | sbrath | ok, will try to turn down the audio on the speaker.. |
| 20:09.37 | sbrath | To go to a business and build the box, place the phones, and train the newb's |
| 20:10.01 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 20:10.03 | sbrath | I guess most of it can be done in advance, and shipped. |
| 20:10.40 | sbrath | I could use a few people who I can pay/per/hour for support help.... |
| 20:10.44 | p3nguin | If the system with running OS is connected to the internet, the rest can be done remotely. |
| 20:11.19 | sbrath | I install openVPN on the boxes so I can also get in and manage the phones. |
| 20:12.34 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v011-006.mobile.uci.edu) |
| 20:13.25 | citywok | okay crazy it dropped again. this time Spawn extension (conference-bridge, s, 3) exited non-zero on 'SIP/1504-0000163c' |
| 20:13.40 | citywok | conference-bridge,s,2 is MeetMe() so s,3 means it exited meetme for some reason |
| 20:18.13 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.9.62) |
| 20:18.18 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.9.62) |
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| 20:36.26 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 20:39.11 | fireman_biff | Hi, with asterisk 1.4 I'm using this command within an extensions_custom.conf: "exten => s,n,Dial(ZAP/g0/5551234,30,r)" When I dial in from my cell and go to that extension, it rings on the other end but as they answer it immediately drops the call. Both the inbound and outbound call are using the same PRI. CLI output is here: http://pastebin.com/CMnxWRPV |
| 20:40.25 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.14) |
| 20:40.29 | WIMPy | fireman_biff: Do you have 2bct enabled, but it's not dupported on your line? |
| 20:41.17 | fireman_biff | WIMPy: Not sure what that is, where would I look? zapata.conf? |
| 20:41.31 | WIMPy | yes |
| 20:42.12 | WIMPy | But I'm not sure what it's called there. |
| 20:42.31 | WIMPy | Watch for anything containign 'transfer'. |
| 20:42.40 | *** join/#asterisk Weazel (~Weazel-@213.8.83.6) |
| 20:43.42 | fireman_biff | a grep shows nothing for "2bct" and the only thing with transfer is "transfer=yes". Could that be what I'm looking for? |
| 20:44.19 | WIMPy | Probably. |
| 20:45.21 | *** part/#asterisk [Outcast] (~anonymous@64.202.62.5) |
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| 20:45.57 | fireman_biff | I'm seeing it referred to in documentation as "flash-hook call transfer".. is that the same as 2bct? |
| 20:46.44 | fireman_biff | alright, looks like that's it |
| 20:46.56 | fireman_biff | I'll try changing it and see how it goes |
| 20:46.58 | fireman_biff | thanks a lot |
| 20:47.58 | WIMPy | No, that's not the same, but it may be the same configuration option. |
| 20:53.22 | fireman_biff | well what I found on voip-info says that the '"transfer" keyword in zapata.conf enables/disables 2BCT on channels', so I'm hoping it will work |
| 20:53.31 | fireman_biff | can't test it for another 1/2 hour though |
| 20:55.18 | fireman_biff | but I'm also seeing a variable called "facilityenable" |
| 20:55.29 | *** join/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 20:55.29 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 20:56.12 | BMJ | Asterisk Developer Conference Call starting at 5:00 PM EST. Info here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Weekly+Developer+Conference+Call |
| 21:02.46 | WIMPy | BMJ: What conf#? I can't seem to find any mention of it. |
| 21:05.19 | BMJ | WIMPy: Pull the ICS file from the wiki page. It's got all the details. |
| 21:06.14 | Qwell | It's also on the child page linked at the bottom |
| 21:07.04 | putnopvut | BMJ: is it not possible to enter the conference bridge yet? I tried and just hear Allison say "That is not a valid conference number. Please try again" |
| 21:07.14 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 21:11.22 | BMJ | putnopvut: Bridge won't open till 4:45 PM EST. |
| 21:12.13 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.23) |
| 21:13.01 | putnopvut | BMJ: thanks |
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| 21:18.23 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.21) |
| 21:20.09 | bjhaid | <PROTECTED> |
| 21:22.01 | carrar | ~book |
| 21:22.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 21:22.30 | carrar | yes |
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| 21:24.49 | bjhaid | <PROTECTED> |
| 21:26.34 | _Corey_ | BMJ: I missed the roadmap session at Astricon... any chance that video is going to make it onto http://www.astricon.net/videoPresentations.aspx ? |
| 21:26.40 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 21:27.31 | BMJ | _Corey_: Asterisk SCF or Asterisk 1.X? |
| 21:28.04 | _Corey_ | 1.x - Wasn't there something on the last day? |
| 21:29.58 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 21:30.17 | malcolmd | discussion by russell on the last day was about new features and changes for 1.8. items that ended up on the 1.10 list were discussed at the astridevcon the day after astricon. details of discussed features including prioritizations are recorded here: https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010 |
| 21:30.22 | BMJ | _Corey_: AstriDevCon. |
| 21:30.34 | BMJ | Thanks, Malcolm |
| 21:30.38 | malcolmd | np :) |
| 21:30.41 | _Corey_ | Ahh... I drove home early and missed that. :) |
| 21:31.19 | JasonL | Hello.. I've been trying to get incoming caller ID working, but no matter what it keeps coming in blank... i use a TDM800P. Any pointers? I've been googling and trying different things but nothing has worked so far |
| 21:31.28 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 21:31.32 | malcolmd | no worries. we were a little worried about some people having to miss it, but it worked out better for the majority to hold that session on the last day, since it also included a brief asterisk scf discussion - and that couldn't have taken place before astricon. ;) |
| 21:31.33 | bjhaid | <PROTECTED> |
| 21:31.50 | malcolmd | JasonL: tried digium support directly? http://www.digium.com/support |
| 21:31.54 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 21:32.11 | JasonL | ah, ok i guess that might help :) thanks malcolmd |
| 21:32.42 | _Corey_ | BMJ: Looks like some case study videos didn't make the cut... *hint* *hint* :) |
| 21:32.49 | malcolmd | JasonL: not quite the answer you're looking for, i know, but it's the best avenue for timely and correct assistance - since the support team deals with these kinds of questions day in and day out :) |
| 21:33.25 | malcolmd | _Corey_: blame the attendees - see photos at the bottom of that wiki page. ;) |
| 21:34.01 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
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| 21:35.15 | BMJ | _Corey_: You can thank the TMC AV team for that. ;-) |
| 21:36.21 | _Corey_ | Yeah, I didn't see any of the other videos from that room. Figures. :) |
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| 21:54.55 | BMJ | Asterisk Developer Conference Call starting at 5:00 PM EST. Bridge now open. Info here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Weekly+Developer+Conference+Call |
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| 22:05.22 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:06.14 | jaytee | i'm trying to block an ipaddress in iptables but it isn't taking effect. I did iptables -A INPUT -s xxx.xxx.xxx.xxx -j DROP |
| 22:06.48 | jaytee | and then I did a save and a restart of iptables but I'm still getting hammered with registration attempts from the ip I want to block |
| 22:06.50 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 22:07.00 | ChannelZ | make sure you don't have some rule preceeding it that would allow it |
| 22:07.11 | ChannelZ | iptables will stop at the first match (in general) |
| 22:07.21 | ChannelZ | iptables -L INPUT -v -n |
| 22:07.54 | ChannelZ | oh.. reg attempts - specify UDP to make sure, I can't remember if saying nothing does both |
| 22:08.21 | jaytee | ChannelZ, I'll give that a shot |
| 22:08.52 | ChannelZ | hmm I just tried a sample here and it says "all" for proto, so that might not be it. Could be an earlier rule |
| 22:09.04 | ChannelZ | pastebin your input table if you want |
| 22:09.42 | Elijah_ | I've always used iptables -A INPUT -s ... -j DROP and it's worked fine |
| 22:10.15 | lanning | iptables -I INPUT -s x.x.x.x -j DROP |
| 22:10.20 | ChannelZ | yeah it's probably matching something earlier in the table like an ACCEPT for port 5060 from anywhere or something |
| 22:10.37 | lanning | iptables -I INPUT 1 -s x.x.x.x -j DROP |
| 22:10.39 | ChannelZ | err ALLOW. whatever |
| 22:10.56 | ChannelZ | yah that'll work too, shove it in the head to be sure |
| 22:23.57 | jaytee | well, I tried it with -p udp and it took a few seconds so it either worked or whoever was trying at that address gave up |
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| 22:24.52 | jaytee | really need to put fail2ban on here but I haven't used it before and don't want to guinea pig a production system. |
| 22:26.37 | jaytee | oh, well day is over and I'm out. might be back later |
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