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| 00:33.51 | telnettech | what does this mean? Jun 1 19:42:55 WARNING[26912]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x2aaab4164170', 10 retries! |
| 00:37.33 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 00:38.16 | jaytee | telnettech, lemme guess Brian, it's on a 1.2 install, right? |
| 00:38.29 | telnettech | lol......yes |
| 00:38.55 | drmessano | wtf |
| 00:39.05 | telnettech | im sorry guys |
| 00:39.22 | telnettech | we are moving as fast as snails into the 21st century |
| 00:39.23 | jaytee | if you search Google ( a handy tool I highly recommend) you'll find lots and lots of hits on this error and they almost always involve 1.2 |
| 00:39.54 | jaytee | the error by itself doesn't mean much. |
| 00:40.28 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 00:40.36 | telnettech | i dont find anything when i do a google search |
| 00:41.05 | jaytee | then your Google-fu is weak, grasshopper |
| 00:41.50 | jaytee | I get over 10 pages of hits when I search for channel_find_locked: Avoided initial deadlock |
| 00:45.10 | telnettech | i do now that i limited what i had in the search bar |
| 00:45.17 | *** join/#asterisk dshap (n=dshap@ip70-181-91-110.oc.oc.cox.net) |
| 00:45.28 | telnettech | so it looks like a know issue with early 1.2 versions |
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| 00:50.35 | dshap | can someone help me with an outgoing call issue i've been having for a couple days now? i unplugged my router and plugged my server directly into the modem and it worked fine, which leads me to believe it is a router/NAT issue |
| 00:50.37 | jaytee | telnettech, does it happen randomly or are you able to reproduce the error? |
| 00:50.49 | dshap | the files i'm using are here: |
| 00:50.51 | dshap | http://pastebin.com/m1241aeb4 |
| 00:50.57 | dshap | oops |
| 00:50.58 | dshap | Call file: http://pastebin.com/m1fb34720 |
| 00:50.58 | dshap | Sip.conf: http://pastebin.com/m5d6b6b11 |
| 00:50.58 | dshap | SIP DEBUG for an outgoing call attempt: http://pastebin.com/m4c96a0a3 |
| 00:51.00 | dshap | there |
| 00:51.07 | telnettech | it happens randomly but you can see the pattern if that makes since |
| 00:51.41 | jaytee | telnettech, if you say there's a pattern I'll take your word for it |
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| 00:51.54 | jaytee | cuz that's really all I've got :-) |
| 00:53.06 | telnettech | so it looks like it is adviseable to upgrade but i can only go to 1.2.28 per our development team's approval |
| 00:57.05 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 00:57.58 | dshap | what might a router/NAT issue be in which outgoing calls are inhibited but incoming calls are allowed? |
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| 01:02.04 | jaytee | ~sipnat |
| 01:02.05 | infobot | [~sipnat] Quick guide on configuring * + 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 |
| 01:02.52 | KyleK | hrm |
| 01:03.16 | KyleK | dshap: can you packet sniff on the router? |
| 01:03.29 | dshap | hm |
| 01:03.32 | KyleK | dshap: personally i've got reinvite=no all around on my * |
| 01:03.49 | dshap | you mean "canreinvite" ? |
| 01:04.16 | KyleK | maybe |
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| 01:05.15 | KyleK | eyup |
| 01:06.11 | dshap | could this have something to do with the fact that my server has a static local IP? |
| 01:06.13 | dshap | 192.168.2.25 |
| 01:06.15 | dshap | within my network |
| 01:07.31 | KyleK | for some reason i have externip=24.x.x.x instead of externhost |
| 01:07.54 | dshap | yea i have externip also |
| 01:08.03 | dshap | it says in that SIP guide that you use externip if you don't use dynamic DNS |
| 01:08.10 | dshap | which i don't |
| 01:08.14 | dshap | and evidently you don't either |
| 01:08.21 | jaytee | externhost according to the sipnat guide is if you use a dynamic DNS service |
| 01:08.21 | dshap | hm |
| 01:08.25 | dshap | yep |
| 01:08.51 | KyleK | well i have a dynamic dns host i just had a little trouble with it |
| 01:09.29 | jaytee | do you have localnet=192.168.2.0/24 ? |
| 01:09.52 | dshap | yes |
| 01:09.59 | dshap | Sip.conf: http://pastebin.com/m5d6b6b11 |
| 01:10.40 | dshap | maybe i should restart my router and see if that helps |
| 01:10.40 | KyleK | i have qualify=yes in my general |
| 01:10.44 | dshap | hm |
| 01:11.31 | dshap | KyleK: adding qualify = yes gave me |
| 01:11.32 | dshap | *CLI> sip s[Jun 1 18:11:09] NOTICE[5677]: chan_sip.c:16223 sip_poke_noanswer: Peer 'flowroute' is now UNREACHABLE! Last qualify: 0 |
| 01:11.33 | dshap | h[Jun 1 18:11:09] NOTICE[5677]: chan_sip.c:16223 sip_poke_noanswer: Peer 'voipms' is now UNREACHABLE! Last qualify: 0 |
| 01:11.36 | jaytee | you have nat=yes in general and then nat=no for each of your VOIP providers. I'm not an expert at using VOIP with ITSPs but that looks wrong |
| 01:12.09 | dshap | jaytee: according to the SIP guide that looks okay |
| 01:12.15 | jaytee | ok |
| 01:12.30 | KyleK | i have nat=yes on my setup though |
| 01:12.34 | *** join/#asterisk BeeBuu (n=beebuu@219.132.191.135) |
| 01:12.46 | MaliutaLap | KyleK: me too, works fine for me |
| 01:13.08 | MaliutaLap | and the conntrack sip helper is making things smoother |
| 01:13.11 | jaytee | KyleK, MaliutaLap, for your ITSP account info? |
| 01:13.21 | jaytee | or in general? |
| 01:13.57 | MaliutaLap | jaytee: well the connections to ITSP's are the only stuff that passes through the nat ... |
| 01:14.00 | KyleK | ugh 22 new messages I really need to make the imap thing work |
| 01:14.31 | jaytee | Maliuta, yeah and that's why I thought the accounts should have it set to yes also. |
| 01:14.48 | jaytee | bet [TK]D-Fender could pinpoint it in less than a minute |
| 01:15.17 | drmessano | KyleK: Good luck with that |
| 01:15.34 | dshap | okay qualify=yes made it so i couldn't receive incoming calls, so i got rid of htat |
| 01:15.34 | dshap | i just tried nat=yes on each of my providers with no luck |
| 01:15.34 | dshap | the "nat" thing confuses me |
| 01:15.34 | dshap | when i plugged my server directly into my modem (no NAT), it was able to work perfectly incoming & outgoing |
| 01:15.34 | dshap | even with nat=yes in my general |
| 01:15.36 | dshap | when i obviously wasn't behind a nat |
| 01:15.37 | KyleK | nat=yes is in general, but I don't mind looping traffic in circles within my lan |
| 01:16.00 | dshap | jaytee: he was helping me last night and wasn't sure but i hadn't narrow the issue down to my router at that point |
| 01:16.12 | dshap | jaytee: although it was a sunday night so he may not have been in his element |
| 01:16.42 | MaliutaLap | jaytee: just checked my sip.conf ... I have nat=no on the phones and nat=yes on the ITSP peers |
| 01:17.35 | jaytee | dshap, what kind of router? |
| 01:17.40 | dshap | belkin |
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| 01:18.03 | dshap | Belkin F5D8230-4 v2 |
| 01:18.24 | dshap | the part that kills me is that i had this working earlier |
| 01:18.28 | dshap | (4 days ago or so) |
| 01:18.30 | dshap | behind the router |
| 01:18.36 | dshap | i know my externIP changed |
| 01:18.40 | dshap | i can't figure out what else did |
| 01:18.57 | dshap | i think i may try restarting my router |
| 01:19.01 | dshap | i'll be back in a few min |
| 01:19.10 | KyleK | beep beep reboot |
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| 01:19.30 | jaytee | if your externIP changed then you need externhost instead |
| 01:19.35 | jaytee | oops, too late |
| 01:20.02 | jaytee | or maybe not |
| 01:20.21 | jaytee | this crap confuses me no matter how much I try to wrap my head around it. |
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| 01:23.22 | dshap | KyleK: router reboot was futile |
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| 01:28.04 | dshap | maybe i should try to set this up with dynamic DNS |
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| 01:33.19 | dshap | oh my god |
| 01:33.22 | dshap | KyleK: i got it working |
| 01:33.34 | dshap | i don't understand this at all |
| 01:33.46 | dshap | on my router setup page I have 2 attributes under "internet settings" |
| 01:33.53 | dshap | one is the WAN IP |
| 01:34.06 | dshap | which is what i *thought* should have gone in my externip |
| 01:34.13 | dshap | the other is Default Gateway |
| 01:34.28 | dshap | which is similar to but not totally the same as my WAN IP |
| 01:34.35 | dshap | i put externip=my default gateway and now it works |
| 01:34.45 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:34.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 01:34.59 | Qwell | dshap: does it end in .1? |
| 01:35.10 | jaytee | dshap, wow, that's odd. the address isn't the same as the external IP? |
| 01:35.11 | dshap | default gateway ends in .1 |
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| 01:35.18 | dshap | wan IP odes not |
| 01:35.19 | Qwell | then you're using the wrong one. |
| 01:35.42 | dshap | what i *thought* i should be using, is the IP that i get from www.whatismyip.com |
| 01:35.42 | johnakabean | hey everyone; i have a problem with asterisk, fresh make. STARTING ASTERISK |
| 01:35.43 | johnakabean | Asterisk ended with exit status 255 |
| 01:35.43 | johnakabean | Asterisk exited on signal 127. |
| 01:35.43 | johnakabean | cat: /var/run/asterisk.pid: No such file or directory |
| 01:35.43 | johnakabean | Automatically restarting Asterisk. |
| 01:35.49 | johnakabean | sorry for flood |
| 01:35.52 | Qwell | dshap: That is the one you should be using. |
| 01:36.04 | dshap | Qwell: well when I use that one, my outgoing calls don't go through |
| 01:36.12 | dshap | Qwell: when i use the default gateway one, they do |
| 01:36.13 | Qwell | then your config is wrong |
| 01:36.31 | dshap | Qwell: would you be willing to take a look and let me know if you see what the problem is? |
| 01:36.32 | johnakabean | asterisk runs perfect when I can get it to stay running |
| 01:36.43 | Qwell | ~nat |
| 01:36.44 | infobot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 01:36.50 | dshap | i read that |
| 01:37.26 | dshap | Qwell: the weird thing is that my problems started when my ISP changed IP addreses for me (dynamic IP) |
| 01:37.36 | dshap | i don't think my configs changed |
| 01:37.42 | dshap | except for externip which i updated |
| 01:37.56 | dshap | is it possible before that my WAN IP was the same as my default gateway? |
| 01:38.02 | dshap | (for the router, i mean) |
| 01:38.18 | dshap | and that after the change, i have different addresses for them, and need the default gateway address for my externip parameter? |
| 01:38.19 | Qwell | no |
| 01:38.23 | dshap | ok |
| 01:38.26 | dshap | well i dont know what's up then |
| 01:38.46 | johnakabean | <johnakabean> Asterisk ended with exit status 255 |
| 01:38.46 | johnakabean | <johnakabean> Asterisk exited on signal 127. |
| 01:38.46 | johnakabean | <johnakabean> cat: /var/run/asterisk.pid: No such file or directory |
| 01:39.02 | johnakabean | running asterisk 1.4.45 or w/e latest is |
| 01:40.10 | johnakabean | 1.4.25 |
| 01:41.06 | johnakabean | is addons 1.4.8 proper for this version? |
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| 01:49.51 | orpheee | hello |
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| 01:50.50 | cp5 | ok guys...this is gonna get weird. two threads, one channel. |
| 01:50.54 | orpheee | how can i choose spécific number on my IPBX, they are a rules about number (example : 7541 or 452 or 12...) ? |
| 01:51.10 | cp5 | what would cause two different threads to have the exact same SIP channel (SIP/whatever-whatever) |
| 01:51.57 | cp5 | that's including the -hex -- once one of them hung up, it caused a seg fault (1.6.0.9) |
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| 02:04.54 | CoffeeIV | I am looking at writing some AGI scripts in python, of the python AGI projects is there one that people use more than others ? All seem semi-abandoned |
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| 02:50.55 | thehar | anyone an ael magician? |
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| 05:50.04 | dshap | for executing simple SQL queries against a database, I would use func_odbc, but for doing more complex/computational stuff I would need to call an AGI script - is this correct? |
| 05:50.17 | dshap | would you ever use AGI for basic database interaction? |
| 05:50.40 | dshap | question was to anyone who cares to comment ^ |
| 05:51.52 | b14ck | dshap--I would, other programming languages make your life easier. |
| 05:52.08 | b14ck | PHP, for example, has a great SQL API for handling databases. |
| 05:52.15 | dshap | hm true |
| 05:52.28 | b14ck | you can do it through asterisk, but it's a bit more painful then is necessary |
| 05:54.21 | dshap | let's say i want to have a database where a phone number is mapped to an audio filename. i want to set up a system where people call in and their callerID is then looked up in the database and the appropriate audio file is played |
| 05:54.52 | b14ck | that would probably best be done using PHP agi |
| 05:54.55 | b14ck | in my opinion |
| 05:55.22 | dshap | would it be okay to put all of the audio files in 1 big directory and then use a PHP AGI call on an incoming CID to get the filename from the database |
| 05:55.27 | dshap | and then play it? |
| 05:56.01 | b14ck | yep |
| 05:56.21 | dshap | would there be a downside to putting them all in 1 directory vs. having a directory set up for each user/callerID? |
| 05:56.25 | dshap | if each user can have multiple audio files |
| 05:56.32 | b14ck | depends on your database scheme |
| 05:56.39 | b14ck | for example, let's say you have 10 caller ids |
| 05:56.56 | b14ck | and each caller id is a separate company, and requires a separate file to be played |
| 05:57.13 | b14ck | i would probably have a subfolder on my system for each company, with the appropriate audio files for each company nested inside |
| 05:57.25 | b14ck | but thats only a matter of organization, you can choose to do it however you want |
| 05:57.30 | dshap | right |
| 05:57.31 | dshap | and then |
| 05:57.37 | KyleK | dshap: crapload of files in a directory is a file system issue |
| 05:57.47 | dshap | what about if the database rows had callerID --> audio file |
| 05:57.54 | KyleK | whats the file system? |
| 05:57.55 | dshap | so if a company had 5 audio files |
| 05:58.13 | dshap | well actually nvm scratch that |
| 05:58.57 | b14ck | dshap, the value for the field callerID would just be the file path to the audio file to play im assuming |
| 05:58.57 | dshap | KyleK: whatever the default CentOS file system is |
| 05:58.57 | b14ck | dshap, that way you can do something like: SELECT callerID from bigdatabase |
| 05:58.57 | b14ck | and get the audio file path into a string |
| 05:59.07 | KyleK | dshap: type "mount" at a prompt ;) |
| 05:59.09 | b14ck | then you can just do like: $agi->stream_file($file) |
| 05:59.14 | b14ck | to play the file to the channel |
| 05:59.23 | dshap | gotcha |
| 06:00.15 | dshap | KyleK: is "ext3" a filesystem type? |
| 06:00.23 | b14ck | yep |
| 06:00.27 | dshap | tha'ts it |
| 06:01.36 | dshap | okay cool i think i know where im goin with this then |
| 06:01.46 | KyleK | wikipedia says ext2 craps out around 10000-15000 files |
| 06:01.52 | dshap | ah |
| 06:02.06 | b14ck | you are most likely using ext3 if you are using centos |
| 06:02.11 | dshap | yeah |
| 06:02.13 | dshap | it says ext3 |
| 06:02.20 | b14ck | And you would need a ton of files to fragment the filesystem. |
| 06:02.22 | b14ck | Don't worry about it. |
| 06:03.25 | dshap | okay b14ck your recommendation for me at this point is to skip the chapter on func_odbc and look at AGI? |
| 06:03.31 | dshap | i already know how to work with PHP |
| 06:03.44 | b14ck | yep, just google asterisk agi, and you'll see that the first link contains all the info you need |
| 06:03.49 | dshap | great |
| 06:03.49 | dshap | thanks |
| 06:04.04 | b14ck | no problem |
| 06:04.08 | dshap | what about AMI? |
| 06:04.17 | dshap | would i be able to do that with PHP as well? |
| 06:04.17 | b14ck | the AMI is another way to communicate with asterisk through sockets |
| 06:04.21 | dshap | right |
| 06:04.22 | b14ck | yep, you would |
| 06:04.27 | b14ck | but the AGI is typically easier to work with |
| 06:04.29 | dshap | let's say i wanted to initiate a call from a web application |
| 06:04.35 | dshap | AMI would be appropriate, not AGI, correct? |
| 06:04.46 | b14ck | yep |
| 06:04.47 | dshap | since AGI is called from wtihin a dialplan |
| 06:04.58 | dshap | okay so i would use those fsock functions to interact with AMI |
| 06:05.04 | b14ck | yes |
| 06:05.14 | dshap | seems like it would be tough/hard to debug |
| 06:05.17 | b14ck | and then you can just send the commands to your socket object |
| 06:05.18 | dshap | doing tht |
| 06:05.25 | b14ck | nope, there are tons of examples out there, very well documented |
| 06:05.30 | b14ck | extremely easy to learn too |
| 06:05.35 | dshap | and it's all with generic PHP socket functions |
| 06:05.36 | b14ck | i got started with it in less than 2 hours |
| 06:05.37 | dshap | nothing asterisk specific |
| 06:05.41 | b14ck | yep! |
| 06:05.47 | dshap | hah alright i'll take your word for it |
| 06:05.51 | dshap | i've got a lot of stuff to learn |
| 06:05.53 | dshap | but it sounds sweet |
| 06:06.08 | b14ck | ya, and feel free to ask questions when you bump into them |
| 06:06.12 | dshap | thank you |
| 06:06.18 | b14ck | this community is all about sharing info ^^ |
| 06:06.23 | dshap | i like it |
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| 07:53.47 | dandre | hello, |
| 07:54.38 | dandre | I am trying to skip the next line if variable FOO is not set. I have tried this: |
| 07:55.22 | dandre | exten => s,n ,GotoIf($["x${FOO}"="x"],n+2) |
| 07:55.34 | dandre | but this doesn't work |
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| 07:56.07 | dandre | is there a simplier way than defining a label? |
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| 08:09.51 | wdoekes | you could use GosubIf instead |
| 08:10.16 | wdoekes | that is not "simpler", but possibly prettier |
| 08:13.06 | dandre | ok |
| 08:13.10 | kaldemar | dandre: your syntax for the gotoif is wrong. after ] should be ?, not ",". |
| 08:13.39 | dandre | yes kaldemar |
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| 08:13.59 | dandre | I had put ? in my file |
| 08:16.49 | kaldemar | and n+2 won't work, $[${PRIORITY} + 2] will give you the priority plus 2. |
| 08:17.43 | dandre | ok, thanks kal |
| 08:18.17 | kaldemar | so GotoIf($["${foo}x" = "x"]?$[${PRIORITY} + 2]) will do it. but i'd still use a label. |
| 08:18.17 | dandre | kaldemar: I just have noticed that +2 works too |
| 08:18.46 | kaldemar | seems to. interesting. |
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| 08:42.06 | pcdog | good morning! |
| 08:42.44 | pcdog | small question: anyone here that uses asterlink and can tell me if there is an outtage or if I simply cannot connect to them from switzerland anymore? |
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| 09:26.27 | OB_Neil | Hi, is there a full list of what events the AMI can send out - I seem to be struggling to find one |
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| 09:32.05 | torrikft | what could be the cause of silence on both ends when doing callback over a SIP trunk? |
| 09:32.21 | torrikft | normal call works fine |
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| 11:03.53 | MadGhost | hello, can I ask questions? |
| 11:04.59 | wackypl | MadGhost ? |
| 11:05.48 | MadGhost | I need SIP server. How different between SER and Asterisk? |
| 11:07.01 | wackypl | i don't know SER ;) |
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| 11:10.20 | kaldemar | very different. SER/Kamailio/OpenSIPS is a proxy, and asterisk is a B2BUA. |
| 11:12.07 | kaldemar | if you could elaborate your needs, someone will probably tell you which one to go for. |
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| 11:24.30 | MadGhost | sorry, I lost connection :-( |
| 11:24.45 | MadGhost | How different between SER and Asterisk? |
| 11:25.02 | MadGhost | Can I using Asterisk how PABX? |
| 11:25.40 | kaldemar | very different. SER/Kamailio/OpenSIPS is a proxy, and asterisk is a B2BUA. |
| 11:26.18 | kaldemar | what's best for you, depends on your needs. |
| 11:26.55 | MadGhost | I need simple PABX for SIP phones in office. |
| 11:27.19 | MadGhost | What I need? |
| 11:27.31 | MadGhost | Sip proxy with IVR or Asterisk? |
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| 11:31.03 | kaldemar | no need for proxy there, only you can decide if you need IVR. asterisk should be good for you. |
| 11:32.12 | MadGhost | thanks :-) |
| 11:33.08 | MadGhost | How much real clients Asterisk can serve? |
| 11:34.14 | MadGhost | How much clients can simultaneously can work through Asterisk? |
| 11:35.15 | kaldemar | depends on used hardware, see some reports here: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
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| 11:54.29 | MadGhost | <kaldemar> thank you |
| 11:54.58 | johnakabean | Anyone know how to set outbound routes based on time conditions? |
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| 11:59.27 | kaldemar | johnakabean: http://www.voip-info.org/wiki/view/Asterisk+tips+openhours |
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| 12:15.45 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:20.47 | HenrikBe | How can I check agent status (logged in/not logged in), I use rawman to show agent status, but no field seem to be reliable for this purpose? |
| 12:24.47 | [TK]D-Fender | HenrikBe: How do your agents "log in"? |
| 12:27.04 | HenrikBe | TK: I am building a login-form in an web-based application, they enter their agentnumber and password, and this is then submited via rawman to originate a agent login call to the extension of the user. |
| 12:27.58 | wackypl | HenrikBe: voipbilling.pl ;) |
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| 12:29.43 | [TK]D-Fender | HenrikBe: AgentLogin()? |
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| 12:30.37 | HenrikBe | TK: yes, I call the users extensions, when it is answered I use AgentLogin to login the agent |
| 12:31.55 | HenrikBe | TK: And it works, but I need a way to see which agents is logged in |
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| 12:32.18 | [TK]D-Fender | HenrikBe: Ok, well there is an AMI command to dumps your queues & agents, and you can use COMMAND "show queue X" to see who's logged in that way and parse it at worst |
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| 13:09.08 | viraptor | is there a way in 1.4 to have a context with entries both in extensions.conf and in mysql? I've got [some-sub] (i,1,Return) ; (_.,2,Return) in extensions, but I also have (exact_number,1,SomeApp) in the database, but every GoSub(some-sub,ext,1) seems to go straight to invalid handler :/ |
| 13:11.23 | [TK]D-Fender | viraptor: AFAIK you can either use "switch" or not at all. One or the other |
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| 13:13.37 | viraptor | [TK]D-Fender: do you have any ideas on how to handle this situation? I need to run only one specific app from the db, but only 1% or so of users will need it, so I'd rather assume that missing row == not enabled |
| 13:14.06 | [TK]D-Fender | viraptor: As I said i don't believe you can mix hard vvs DB |
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| 13:15.40 | infernix | Qwell: ping? |
| 13:15.45 | *** join/#asterisk jmkgreen (n=chatzill@fentech.gotadsl.co.uk) |
| 13:17.16 | jmkgreen | We are running * 1.4.23.1 making lots of outbound calls with VoiceXML. When this exists, we enter the h part of the dialplan which in turn uses DeadAGI to clean up our database. |
| 13:17.51 | jmkgreen | Problem is, sometimes and apparently at random, this process seizes up, causing lots of hung channels with DeadAGI. |
| 13:18.46 | jmkgreen | The net result is no more calls being placed, and VoiceGlue (our VoiceXML browser) loses it's file handle (presumably a socket to talk to asterisk) |
| 13:19.18 | jmkgreen | Is this a known issue? |
| 13:24.22 | dandre | hello, |
| 13:24.47 | dandre | How can I display CID information on an analog phone connected to a TDM410P ? |
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| 13:28.21 | *** join/#asterisk acehunky (n=acehunky@123.252.144.92) |
| 13:28.27 | acehunky | hello |
| 13:28.42 | [TK]D-Fender | jmkgreen: Issue? For you yes, but its to be expected. Being in "h" * has no reason to terminate the "dead" end of this call short of running out of things to do. Thins means YOU have to monitory for dead processes. |
| 13:29.04 | acehunky | i have a question regarding the G729 license available from Digium ... want to know what kind of processor would i need to be able to transcode around 16E1 worth of calls |
| 13:29.05 | [TK]D-Fender | dandre: "usecallerid=yes" |
| 13:29.10 | virtualme123 | Has anyone had problems with running DeadAGI in the h context and getting stuck in a channel Up state effectively crashing the channel? I trying to understand what might be stopping it from closing the channel down once the script has completed. |
| 13:29.19 | acehunky | around 480 calls ... |
| 13:29.22 | [TK]D-Fender | acehunky: Holy-f'n-shit |
| 13:29.34 | [TK]D-Fender | acehunky: HUGE load... |
| 13:29.42 | [TK]D-Fender | acehunky: Call Digium directly |
| 13:29.48 | [TK]D-Fender | acehunky: This warrants is. |
| 13:29.55 | [TK]D-Fender | it* |
| 13:29.55 | acehunky | [TK]D-Fender: yeah ... thats like SS7 trunks :) |
| 13:30.17 | acehunky | i dont have an option for putting in the TC400B card ... |
| 13:30.25 | acehunky | dont have enough PCI slots :( |
| 13:30.53 | acehunky | also modern servers charge extra for Old PCI technology ... and TC400 doesnt come in PCIe :( |
| 13:31.06 | coppice | I wonder what the greatest number of TC400B cards in one box is? :-\ |
| 13:31.20 | acehunky | yeah steve i aint sure |
| 13:31.26 | dandre | [TK]D-Fender: it is already set |
| 13:31.38 | acehunky | but the last time i got one .. my server used to reboot if i remove the TC400 server becomes stable |
| 13:32.09 | acehunky | dunno when Digium is gonna re-engineer their TC400 with PCI Express form factor |
| 13:32.20 | *** join/#asterisk theron (n=theron@216.51.246.211) |
| 13:32.26 | acehunky | atleast we can custom make some servers having more PCI Express slots |
| 13:32.31 | [TK]D-Fender | acehunky: Only real option I know is very pricy (AudioCodes Mediant 2000 to do the PRI+G.729) |
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| 13:33.05 | acehunky | yeah but on the other end i got SS7 not PRI :( |
| 13:33.32 | dandre | callerid works fine on fxo but not on fxs |
| 13:33.38 | viraptor | [TK]D-Fender: it works after all - as long as switch => Realtime/@ is after the local extensions :) - that part was confusing |
| 13:34.13 | jmkgreen | [TK]D-Fender: So when asterisk reaches the end of the h section of the dialplan, it does not automatically terminate the channel, but leaves it hanging around? I expected a clean exit..? |
| 13:34.14 | [TK]D-Fender | dandre: should work fine |
| 13:34.41 | [TK]D-Fender | jmkgreen: It should terminate at the end of "h" but you alluded that your script didn't |
| 13:34.46 | hesco | dialplan reload gives me in the console: "No category context for line 6 of /etc/asterisk/extensions.conf", which reads: "CID=7702505192", although this syntax parrallels the assignment of other constants in other files included in extensions.conf. Can anyone please advise what I may be missing here, please? |
| 13:35.42 | *** join/#asterisk stew (i=1413@freenode/staff/stew) |
| 13:35.56 | jmkgreen | [TK]D-Fender: Our VoiceXML simply calls <exit/> which completes the call session and hands control back to the browser to finish any housekeeping it needs. |
| 13:36.09 | jmkgreen | We assumed this was the clean way of completing "our world" |
| 13:36.23 | [TK]D-Fender | jmkgreen: "browser"?! |
| 13:36.33 | jmkgreen | I should add this sequence works in over 99% of calls |
| 13:36.38 | jmkgreen | VoiceXML browser |
| 13:36.45 | jmkgreen | called via AGI |
| 13:36.56 | [TK]D-Fender | jmkgreen: If that hangs, you're FUBAR'd |
| 13:37.27 | coppice | hanging FUBARs most people |
| 13:37.43 | hesco | never mind, I moved it into a [global] context and it seems to reload now |
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| 13:37.54 | jmkgreen | yeah that I gathered, quite what we are doing / or not doing to get FUBARed is the question |
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| 13:38.03 | johnakabean | [Jun 2 08:55:39] WARNING[12771] pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/agc.call: Permission denied, deleting |
| 13:38.04 | johnakabean | [Jun 2 08:55:39] WARNING[12771] pbx_spool.c: Failed to scan service '/var/spool/asterisk/outgoing/agc.call' |
| 13:38.14 | [TK]D-Fender | coppice: Short drop and a suddens top :) |
| 13:38.49 | johnakabean | I have chowned to asterisk:asterisk and chmodded 777 BOTH COMMANDS RECURSIVELY the directory /var/spool/asterisk/outgoing AFTER moving the files there |
| 13:38.53 | [TK]D-Fender | jmkgreen: Well if your process never exits, * won't hang up. I'd run a monitoring script if I were you. |
| 13:39.13 | johnakabean | this problem didn't start until upgrade to 1.4.25 |
| 13:39.20 | coppice | [TK]D-fender this is how hanging up on telemarketers should be |
| 13:39.55 | [TK]D-Fender | coppice: I miss the good 'ole days of sending 10,000 volts down the line and sautee-ing them ;) |
| 13:40.28 | coppice | sauteing's too good for 'em |
| 13:40.39 | jmkgreen | [TK]D-Fender: That makes sense. Except I have tcpdumps proving in those particular calls that have hung, we sent VoiceGlue an <exit/> as normal. Which reduces our options of what to look at next. |
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| 13:41.20 | [TK]D-Fender | jmkgreen: If you don't see the dialplan continue, then your script never fully exits. Just make a monitoring daemon and track them. |
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| 13:51.40 | dandre | How can I "see" what cid informations are sent to a fxs channel of a tdm800? |
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| 13:57.13 | rue_more | dandre, |
| 13:57.36 | rue_more | you can use a nop in the dialplan with the cid data as a paramiter |
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| 14:04.45 | dandre | rue_more: the dialplan is ok but the cid is not displayed on the phone attached |
| 14:05.24 | rue_more | is caller id set up on the dahdi channel? |
| 14:07.41 | dandre | I use zaptel channel |
| 14:07.56 | dandre | and usercallerid = yes in eapata.conf |
| 14:08.18 | dandre | all works fine for fxo but not for fxs |
| 14:08.21 | [TK]D-Fender | dandre: Show us |
| 14:09.02 | rue_more | hmm zaptel eh |
| 14:09.11 | rue_more | eapata.conf? |
| 14:09.23 | rue_more | zaptel.conf? |
| 14:09.27 | dandre | sorry, zapata.conf |
| 14:09.44 | rue_more | hmm too early , didn't see that one |
| 14:12.25 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 14:13.11 | dandre | rue_more, [TK]D-Fender: http://pastebin.fr/4670 |
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| 14:15.11 | [TK]D-Fender | dandre: Should work fine |
| 14:15.19 | [TK]D-Fender | dandre: Test your phone elsewhere |
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| 14:32.15 | dandre | [TK]D-Fender: I have tested the phone elsewhere and I get correct cidnum |
| 14:32.35 | [TK]D-Fender | dandre: No idea then.... |
| 14:33.09 | dandre | how can I get debug info from zaptel module? |
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| 14:52.29 | wdoekes | hi there.. I have an issue with call forwarding. namely that I can't get the accountcode (or some other identifier) of the forwarding phone. |
| 14:52.55 | wdoekes | (1) in my incoming context, I call (for instance): Dial(SIP/201&SIP/202) |
| 14:53.36 | wdoekes | (2) if SIP/201 has call forwarding enabled, I get to Local/<outbound-number>@outbound |
| 14:53.41 | [TK]D-Fender | wdoekes: Set a variable to be inherited and re-assign it in your dialplan if its frmo a forward. |
| 14:54.06 | wdoekes | (3) in [outbound] I don't know if it was 201 or 202 that did the forwarding |
| 14:55.21 | wdoekes | [TK]D-Fender: does your answer apply to my question after (3)? |
| 14:55.47 | [TK]D-Fender | wdoekes: Not sure how to detect that. |
| 14:55.52 | *** join/#asterisk just110 (n=root@122.169.79.177) |
| 14:56.04 | wdoekes | in app_dial.c I don't see anything that can help, so I'm tempted to believe it's not possible |
| 14:56.26 | just110 | hello ..... guys, i need a help |
| 14:56.35 | just110 | i am facing strange issue |
| 14:56.46 | wdoekes | after ast_channel_inherit_variables(in, o->chan); I would've liked something like copy(c->accountcode, o->chan->accountcode) |
| 14:57.04 | wdoekes | (which I'm guessing would solve my problem) |
| 14:57.28 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:57.39 | just110 | one side my call is making progress |
| 14:57.52 | VaGoNeTaS | somebody knows how to disable DTMF? |
| 14:58.14 | wdoekes | (if I knew the original channel name, I might alternatively be able to use ImportVar or something (if it still exists)) |
| 14:58.16 | just110 | and at the same time retransmission of INVITE is send |
| 14:59.22 | hfx-ed | I'm making the jump from zap to dahdi. A few gotchas there. ;) I thought I got through them all. Calls are processing, however I am now hearing crackiling very reularly in my audio. |
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| 15:00.32 | wdoekes | tnx for thinking with me at least, [TK]D-Fender. I'll go ask in -dev |
| 15:01.07 | [TK]D-Fender | wdoekes: There are a lot of places I'd like to see hooks made into, this is jsut another practical one... |
| 15:01.22 | just110 | hello fender |
| 15:01.28 | hfx-ed | I am using a TE405P. Was using version 1.4.11, upgraded to 1.4.25. |
| 15:01.34 | just110 | plz help me in my issue |
| 15:02.16 | *** join/#asterisk psilikon (n=psilikon@220-241.187-72.tampabay.res.rr.com) |
| 15:02.39 | [TK]D-Fender | just110: Don't go targeting people an expecting personalized help. It rather rude. |
| 15:02.49 | [TK]D-Fender | just110: And you have shown us nothing. |
| 15:03.08 | hfx-ed | Might anyone know of any parameters in the newer version of * that I should look at? |
| 15:03.34 | just110 | i am sorry...fender |
| 15:03.48 | [TK]D-Fender | hfx-ed: nothing that should involve chrackling. pastebin "dahdi_cfg -vvvv" and "cat /proc/interrupts" |
| 15:05.13 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
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| 15:06.48 | shareenergy | Hello ppl |
| 15:06.57 | shareenergy | I believe round robin has a bug |
| 15:07.18 | shareenergy | if I have 10 agents picking up calls |
| 15:07.24 | shareenergy | and 11 logins |
| 15:07.39 | shareenergy | this 11 gets the calls first then the other 10 |
| 15:08.00 | *** join/#asterisk plq (n=plq@88.250.169.4) |
| 15:08.30 | just110 | hello guys... |
| 15:08.44 | just110 | i am facing strange kind of issue |
| 15:08.59 | just110 | asterisk is sending reinvite |
| 15:09.07 | just110 | 6 times |
| 15:09.24 | [TK]D-Fender | justPlease don't simply repeat the same question every 10 minutes. You're going to ahve to SHOW US the problem. |
| 15:09.26 | just110 | and at the same time that call is doing progress |
| 15:09.41 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
| 15:10.44 | just110 | call is answerd. then also retransmission of invite is continue |
| 15:11.01 | hfx-ed | [TK]D-Fender - The files are at dahdi_cfg http://pastebin.ca/1444944 proc/interrupts http://pastebin.ca/1444949 |
| 15:11.09 | hfx-ed | I notice that echocan is being set to mg2. Previously I had the echo can off as I am using Ditech externally. I wonder if that is where the problem may lie? Thoughts? |
| 15:11.59 | [TK]D-Fender | hfx-ed: Very po it if you are using that external unit |
| 15:12.03 | [TK]D-Fender | possible* |
| 15:14.49 | *** join/#asterisk lancey (i=lancey@support.net1.cc) |
| 15:15.30 | lancey | hi guys, i need help with internal_timing. I have it enabled in asterisk.conf, have ztdummy loaded and properly showing when doing a zap show status, but asterisk still sends the silenceSupp:off header in sip conversations. Anyone any clues? |
| 15:15.46 | lancey | according to the source, it shouldn't be doing so... |
| 15:16.35 | telnettech | any reason why you would have Answer pickup the call before ringing a SIP device? |
| 15:17.45 | telnettech | besides if there was a recording playing in between the 2 dialplan lines |
| 15:17.48 | jaytee | lancey, check the settings for silence suppression on the SIP phones themselves |
| 15:18.44 | lancey | jaytee: it's that the asterisk does generate this in the headers it sents |
| 15:19.03 | lancey | and one telco that we connect to doesn't like it (shit huaweis) |
| 15:19.14 | *** join/#asterisk aenaus (n=hdgfghf@79.107.177.227) |
| 15:19.14 | [TK]D-Fender | jaytee: If telco VM gets in the way. Double-ended timeouts, etc |
| 15:19.55 | lancey | the thing is, asterisk has ztdummy available for timing, has internal_timing = yes in asterisk.conf, yet it generates that header |
| 15:19.57 | *** join/#asterisk kpettit (n=keith@99-172-37-26.lightspeed.tblltx.sbcglobal.net) |
| 15:21.06 | lancey | if (!p->owner || !ast_internal_timing_enabled(p->owner)) |
| 15:21.06 | lancey | <PROTECTED> |
| 15:21.13 | lancey | that's what the sources say. |
| 15:21.50 | lancey | Description Alarms IRQ bpviol CRC4 |
| 15:21.51 | lancey | ZTDUMMY/1 1 UNCONFIGUR 0 0 0 |
| 15:22.00 | lancey | that does mean everything is fine with ztdummy, right? |
| 15:22.21 | lancey | zttest from the command line also works fine... i'm lost :/ |
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| 15:32.56 | *** join/#asterisk AlmightyOatmeal (n=jamie@24-196-139-57.dhcp.mdsn.wi.charter.com) |
| 15:33.39 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:34.01 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:34.10 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 15:34.29 | telnettech | any reason why you would have Answer pickup the call before ringing a SIP device besides if there was a recording playing in between the 2 dialplan lines? |
| 15:35.50 | AlmightyOatmeal | sounds like you answered that |
| 15:36.42 | lancey | telnettech :sometimes you won't get the audio if the channel is not answered |
| 15:37.24 | telnettech | lancey: funny you say that cause with Answer I am not getting the audio for some incoming calls |
| 15:38.01 | telnettech | it is ringing both of my sip devices and when answered by 1of them It is a dead call and the other phone continues to ring |
| 15:38.36 | telnettech | but it doesnt happen for all calls it is just sporadic |
| 15:39.28 | kpettit | I'm using Asterisk 1.6.1, the generic Asterisk recordings are playing with alot of static. Phone calls, and recording message are all fine, but all the default Asterisk recording have alot of static. Wasn't sure if there was a setting or something I needed to toggel to adjust that |
| 15:40.37 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:40.58 | KyleK | like static=no somewhere? |
| 15:41.09 | jaytee | hehe |
| 15:41.16 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 15:41.29 | lancey | telnettech: no, i say if you don't answer, you may not get sound |
| 15:41.34 | kpettit | haha. I'm not sure what it is, haven't had this problem before. It's like the volume for the recordings isn't right |
| 15:42.22 | telnettech | lancey: ha ha |
| 15:42.24 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 15:42.41 | telnettech | Im talking about the Answer function in dialplan |
| 15:42.55 | lancey | telnettech: exactly. |
| 15:43.03 | KyleK | kpettit: maybe you changed gain settings somewhere? |
| 15:43.10 | lancey | you mentioned some recording played. whatever.. never mind. |
| 15:43.19 | jaytee | kpettit, if recorded messages play fine but the standard sound files are playing with static I'd make sure you're not transcoding from one format to another and check your enabled codecs |
| 15:43.50 | VaGoNeTaS | DOES somebody knows howto DISABLE DTMF? |
| 15:44.09 | kpettit | jaytee: AHhh, that's probally it. |
| 15:44.34 | *** part/#asterisk juanIMP (n=Juancho@200.71.41.254) |
| 15:44.35 | kpettit | I'll try them all uncompressed and see what it sound like |
| 15:44.54 | jaytee | kpettit, check the format of the sound files in /var/lib/asterisk/sounds. are they gsm? |
| 15:45.06 | *** join/#asterisk lesouvage (n=lesouvag@92.65.174.153) |
| 15:45.21 | kpettit | jaytee: yes, they are all .gsm |
| 15:45.44 | jaytee | and are your phones enabled to use the gsm codec? |
| 15:46.20 | kpettit | I get the static dialing from a outside phone into the PBX, or using a sip phone. Not using real hardware phones yet. |
| 15:47.12 | VaGoNeTaS | no one knows? |
| 15:47.36 | [TK]D-Fender | VaGoNeTaS: Why would anyone do that? |
| 15:47.42 | kb3ien | Anyone used AgentCallbackLogin ? |
| 15:48.08 | jaytee | kpettit, what codecs are allowed in sip.conf for the sip phone, or if you prefer to try something else you can get the sound files in ulaw or wav formats here: |
| 15:48.11 | jaytee | http://downloads.asterisk.org/pub/telephony/sounds/ |
| 15:48.53 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:49.15 | [TK]D-Fender | kpettit: If your stock recordings sound bad, and others you ahve don't then you likely have : |
| 15:49.17 | jaytee | kb3ien, yes I've used it but it's not reliable and has been deprecated. it's not in 1.6 |
| 15:49.17 | [TK]D-Fender | ~gsmbug |
| 15:49.18 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 15:49.18 | AlmightyOatmeal | i'm new to asterisk and just finally managed to get the gui running.. just wondering where i can find the login/passwd to use (though i did specify secret in manager.conf) |
| 15:49.19 | [TK]D-Fender | ^^^^6 |
| 15:49.24 | kpettit | jaytee: ok, will do. I haven't messed with the codecs on this install yet so I'm not sure what it is. I'll set it to ulaw though to test |
| 15:49.42 | [TK]D-Fender | AlmightyOatmeal: GUI's are not supported in this channel. please see the /topic for other channels |
| 15:49.53 | jaytee | kpettit, and download the ulaw sound files and unpack them to the /var/lib/asterisk/sounds directory then |
| 15:50.11 | kpettit | jaytee: Good idea. |
| 15:50.16 | AlmightyOatmeal | [TK]D-Fender: i figured the login credentials would have been more of an asterisk thing, but i've already asked in the other channel |
| 15:50.29 | kpettit | [TK]D-Fender: thanks for the suggestion, this test shoudl let me know |
| 15:50.32 | [TK]D-Fender | AlmightyOatmeal: then await your answer there |
| 15:50.59 | AlmightyOatmeal | very well |
| 15:51.33 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 15:53.10 | *** join/#asterisk bbryant1 (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 15:53.30 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:56.15 | AlmightyOatmeal | after making a modification to manager.conf is it necessary to restart asterisk completely? |
| 15:56.34 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
| 15:56.35 | jaytee | nope |
| 15:56.54 | leifmadsen | manager reload |
| 15:57.06 | AlmightyOatmeal | ty |
| 15:57.30 | *** join/#asterisk oej (n=olle@192.36.80.8) |
| 15:57.38 | leifmadsen | anyone use 1.6.2 have an issue with the Background() application muting audio after it receives DTMF? |
| 15:57.47 | kpettit | jaytee: ahhhh sound quality much better. ulaw sound files did the trick |
| 15:57.48 | leifmadsen | (might not be specific to the Background() app) |
| 15:57.56 | *** join/#asterisk pcdog (n=pcdog@213.144.146.5) |
| 15:58.31 | jaytee | kpettit, it's probably the gsmbug that [TK]D-Fender pointed you to that was causing it. Unless I'm starved for bandwidth I stick with ulaw or wav formats. |
| 15:58.57 | kpettit | jaytee: ok, good idea. So shoudl I avoid doing voicemail and such in gsm? |
| 15:59.03 | [TK]D-Fender | jaytee: I jsut compile mine properly ;) |
| 15:59.24 | [TK]D-Fender | kpettit: Just do it right, but you should never waste CPU on needless transcoding anyway |
| 15:59.32 | jaytee | kpettit, that depends on your situation. |
| 15:59.32 | kpettit | Mine was whatever package ubuntu uses. I didn't compile it myself. |
| 15:59.40 | AlmightyOatmeal | is there a recomended sip trunk provider that is cheap that offers basic services for a beginner to make/recieve calls to landline phones? |
| 15:59.42 | jaytee | there's the rub! |
| 15:59.57 | leifmadsen | uhhhhhhhhh.... wow this is new |
| 16:00.03 | leifmadsen | one-way audio over a PRI.... |
| 16:00.05 | [TK]D-Fender | AlmightyOatmeal: Same as those for advanced user ;) |
| 16:00.09 | [TK]D-Fender | ~itsplist-us |
| 16:00.10 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 16:00.18 | AlmightyOatmeal | ty |
| 16:00.23 | Chainsaw | AlmightyOatmeal: cheap, helpful, reliable. |
| 16:00.25 | Chainsaw | AlmightyOatmeal: Pick any two. |
| 16:00.25 | kpettit | I'm trying to keep everything uncompressed. So keeping everything ulaw should make life easier |
| 16:00.25 | *** join/#asterisk icyValk77 (n=icyValk7@host86-161-124-210.range86-161.btcentralplus.com) |
| 16:00.33 | AlmightyOatmeal | Chainsaw: :) |
| 16:00.40 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
| 16:00.46 | jaytee | leifmadsen, one way inbound or outbound? |
| 16:00.53 | leifmadsen | jaytee: outbound |
| 16:01.01 | KyleK | ~itsplist-ca |
| 16:01.02 | infobot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
| 16:01.21 | leifmadsen | call comes in, get audio, then hit Background(), press '5', then audio goes away when listening to the prompt. Calls a phone, then audio comes back from phone -> cell |
| 16:01.38 | leifmadsen | but not cell --> phone |
| 16:01.44 | leifmadsen | (phone is defined as a SIP extension) |
| 16:01.48 | leifmadsen | (on the LAN) |
| 16:02.11 | jaytee | leifmadsen, this 1.6? |
| 16:02.32 | kb3ien | is there something like agentlogin for 1.4 that does not make the agent sit on hold? |
| 16:02.42 | leifmadsen | jaytee: yes, 1.6.2 branch (latest) |
| 16:03.03 | leifmadsen | kb3ien: yes -- AddQueueMember() application. You build the agent callback login stuff in the dailplan |
| 16:03.05 | jaytee | leifmadsen, I've never seen that before on earlier version of 1.6 or 1.4, must be a bug |
| 16:03.37 | leifmadsen | jaytee: ya... I'm going to try using an earlier 1.6 and see if it is asterisk itself, or something to do with the wanpipe stuff |
| 16:04.01 | kb3ien | leifmadsen is it any more stable? |
| 16:04.16 | leifmadsen | kb3ien: in relation to what? |
| 16:04.52 | jaytee | i've seen where in 1.4.22 on my IVR server where Background would drop audio outbound and I'd have to restart but I'd chocked it up to a bad install of Lumenvox that was causing all kinds of issues. |
| 16:05.05 | outtolunc | notes the amount of lint in my pocket seems to be stable |
| 16:05.11 | leifmadsen | jaytee: ya, I'm going to roll back to 1.6.0 and see if I get the same issue |
| 16:05.20 | kb3ien | leifmadsen that the other AgentLoginCallback stuff? |
| 16:05.38 | jaytee | kb3ien, it's not deprecated so it will work in 1.6 if you move to 1.6 |
| 16:05.44 | leifmadsen | kb3ien: since AgentLoginCallback() no longer exists, and that it's "just dialplan", I would suggest yes. |
| 16:05.50 | jaytee | and is the new recommended method |
| 16:06.03 | leifmadsen | there is a script in the 'doc' directory of your asterisk source I believe |
| 16:06.14 | leifmadsen | although I do plan on checking on it, updating it (if necessary) and writing some documentation |
| 16:06.24 | click | hm |
| 16:07.06 | click | leifmadsen: which ports does * require for video-transmissions between two x-lite clients? |
| 16:07.12 | leifmadsen | zero idea |
| 16:07.21 | leifmadsen | the ones that work? |
| 16:07.22 | kb3ien | okay. makes sense. |
| 16:07.23 | click | *ponder* |
| 16:07.58 | click | i'll sniff around |
| 16:08.01 | AlmightyOatmeal | anyone used CallCentric as a sip provider? seems to be decent with $19.95/mo north american unlimited and no setup cost |
| 16:08.01 | leifmadsen | jaytee: oh nice -- 1.6.0 does not have this issue |
| 16:08.43 | AlmightyOatmeal | those setup costs are pretty steep heh |
| 16:08.47 | click | almightyoatmeal: ordered BroadVoice here, 14.95 and unmetered free calls to 21 countries - satisfied |
| 16:08.51 | jaytee | leifmadsen, not surprising. There's been so many changes between the two although 1.6.2 looks very promising when it gets stable. |
| 16:08.57 | click | or was it 19.95... |
| 16:09.04 | leifmadsen | jaytee: checking 1.6.1 now to see if the regression has carried through there |
| 16:09.30 | AlmightyOatmeal | click: that was going to be my second choice, but $39.95 activation is just wowzers |
| 16:09.39 | AlmightyOatmeal | i'm just a poor kid learning asterisk hehehe :) |
| 16:09.42 | jaytee | leifmadsen, my gut says it found it's way into the code in 1.6.2 and it won't be in 1.6.1 |
| 16:09.42 | click | the activation is just a first-timer anyway |
| 16:09.48 | AlmightyOatmeal | yeah |
| 16:09.49 | click | ah, well, then it's steep |
| 16:09.50 | leifmadsen | jaytee: mine too |
| 16:10.08 | jaytee | speaking of guts, it's lunchtime!!! bbiab |
| 16:10.18 | click | almightyoatmeal: BYOD ? |
| 16:10.24 | AlmightyOatmeal | BYOD? |
| 16:10.27 | click | http://broadvoice.com/rateplans_byod.html |
| 16:10.32 | click | bring your own device |
| 16:10.51 | click | came down to 5.95 + setup, then 11.42 per month after that |
| 16:11.03 | AlmightyOatmeal | oh, neat |
| 16:11.09 | click | eventually with the unlimited world package on top of it |
| 16:11.27 | click | or instead of the monthly-fee thingie |
| 16:11.35 | click | works for US-US/CA at least |
| 16:12.13 | AlmightyOatmeal | nice |
| 16:12.39 | click | (i'm in norway, so it's ... a tad more expensive, but i needed free unmetered calls to US/CA residents) |
| 16:12.50 | AlmightyOatmeal | gives me something to save my allowance for :) |
| 16:13.18 | click | i'm just happy as long as it works |
| 16:13.22 | jameswf | sipgate has free did's right now |
| 16:14.37 | AlmightyOatmeal | jameswf: terminating trunk or just to other voip boxes? |
| 16:15.03 | click | jameswf: well, they still charge for the calls, compared to non-charged with BV |
| 16:15.30 | AlmightyOatmeal | but /window 7 |
| 16:15.33 | AlmightyOatmeal | oops |
| 16:15.42 | kb3ien | i'm running sip show queues and getting now queues, can i define them in queues.conf as [myqueue] ? |
| 16:16.07 | kb3ien | s/now/no/ |
| 16:18.31 | leifmadsen | kb3ien: queue show, not sip show queues |
| 16:19.10 | kb3ien | No queues. |
| 16:19.53 | *** join/#asterisk stijnbe (n=stijnbe@213.49.145.151) |
| 16:20.17 | SuPrSluG | kb3ien:yes. there and agents is where you deifine queues. |
| 16:20.44 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 16:21.23 | kb3ien | looks like a borken #include ... |
| 16:21.28 | SuPrSluG | kb3ien:you should read the wiki or the book for info on queues |
| 16:22.51 | hfx-ed | I've removed the echo can from the situation |
| 16:23.09 | kb3ien | not sure why the #include failed. ill worry later. |
| 16:23.43 | hfx-ed | I set up a test number to play 'all-your-base' via the TE405 to the PRI, and can hear minor crackling in the audio |
| 16:24.08 | hfx-ed | calls that I pass through to other asterisk boex via IAX are having severe crackling |
| 16:24.26 | hfx-ed | clearly the issue is in my gateway |
| 16:24.57 | hfx-ed | for those not hear earlier I've just upgraded from 1.4.11 to 1.4.25 |
| 16:25.12 | hfx-ed | this is when the audio issue started |
| 16:25.42 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 16:25.56 | Kobaz | how can i tell if a call is being transfered in the dialplan... ie: phone a calls phone b... phone b transfers to c.... when the call from b to c takes place... i want to see if this is a transfereed call |
| 16:25.59 | hfx-ed | not sure if there are some parameters from the earlier configuration that did not make it into the newr |
| 16:26.36 | hfx-ed | thoughts? |
| 16:28.59 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 16:36.01 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-46.lsanca.dsl-w.verizon.net) |
| 16:36.07 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 16:38.06 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 16:39.26 | *** join/#asterisk jthurman42 (n=jthurman@66.154.131.25) |
| 16:44.44 | lost_soul | no help here, sorry.. new to asterisk |
| 16:45.33 | jthurman42 | Has anyone here had issues with Cisco 79xx phones calling Voicemail and randomly getting "Maximum retries exceeded" errors? |
| 16:45.35 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:52.23 | *** join/#asterisk jtodd (i=gmbje5uz@ns.fox-den.com) |
| 16:52.23 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:53.18 | nny_1 | hmm. Need a second eye on this issue. I have a script that reloads moh from an internal web page. (I would use AMI for it, but it worked previous ). My webserver runs as "apache" and is a member of the group asterisk. I checked the asterisk.ctl file and added group write to it, but the script still fails when running from the web page. Anything I could have overlooked? |
| 16:53.21 | [TK]D-Fender | Kobaz: go read the CHANNELVARIABLES doc |
| 16:55.48 | *** join/#asterisk VoipForces (n=kvirc@mail.viatransint.com) |
| 16:56.21 | VoipForces | Anyone familiar in getting Audiocode MxP124 FXS to have MWI working ? |
| 16:57.56 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
| 16:59.42 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:59.49 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 17:01.35 | kb3ien | does UnpauseQueueMember change a agent that is Unavailable ? |
| 17:02.36 | leifmadsen | no |
| 17:02.40 | [TK]D-Fender | kb3ien: No, it changes one that is PAUSED |
| 17:02.41 | leifmadsen | PauseQueueMember() does |
| 17:02.55 | leifmadsen | wait, I read that wrong :) |
| 17:03.01 | leifmadsen | what [TK]D-Fender said |
| 17:03.14 | leifmadsen | unavailable != paused |
| 17:03.16 | kb3ien | okay unavailable is a hard state to modify by script. |
| 17:03.39 | kb3ien | i'll make all agents dynamic it seems easier to manage that way. |
| 17:03.53 | [TK]D-Fender | kb3ien: Maybe you should look at why the device/member is "unavailable" |
| 17:04.07 | leifmadsen | like... defining a hint perhaps? |
| 17:04.26 | leifmadsen | in the context you have defined for subscriptions in sip.conf |
| 17:04.39 | leifmadsen | exten => 0004F2040808,hint,SIP/0004F2040808 |
| 17:06.42 | Kobaz | do de do |
| 17:07.01 | Kobaz | is there a way to detect a call is being transfered at the dialplan level |
| 17:09.39 | *** part/#asterisk click (i=click@ti0127a340-0847.bb.online.no) |
| 17:11.00 | nny_1 | is there a sanctioned way to give the asterisk.ctl run file group write permissions or should I just hack the init script? |
| 17:12.33 | leifmadsen | probably asterisk.conf |
| 17:13.29 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 17:13.40 | [TK]D-Fender | Kobaz: Already answered you |
| 17:13.52 | Kobaz | oh |
| 17:13.54 | Kobaz | you did |
| 17:13.56 | Kobaz | hmm, |
| 17:14.02 | Kobaz | oooh |
| 17:14.07 | Kobaz | i didn't see that, it scrolled off |
| 17:14.52 | jaytee | channelvariables,doc is sooo rich, moist and chewy. how can any Asterisk user pass it by? |
| 17:15.00 | Kobaz | okay i see BLINDTRANSFER |
| 17:15.08 | Kobaz | is there a way to catch non-blind transfers |
| 17:15.12 | Maliuta | hmm, geeks and their insomnia ... tonights project - get the luxman up and running. Done |
| 17:15.36 | Maliuta | is slightly concerned about the rest of his parents vinyl collection |
| 17:16.05 | leifmadsen | Kobaz: attended transfers with SIP are not possible to detect because they just look like another phone call until the bridging is done -- which has nothing to do with dialplans. |
| 17:16.21 | leifmadsen | Kobaz: may be possible if you're using built in transfers with the * keys in features.conf |
| 17:16.23 | Kobaz | leifmadsen: yeah, i was afraid of that |
| 17:16.39 | leifmadsen | Kobaz: it is not possible. |
| 17:16.43 | jaytee | frankly I miss the hiss, static and pop of LPs. CD's and mp3's just don't have the "ambience" that vinyl did. |
| 17:16.45 | leifmadsen | there is no dialplan being executed to related them |
| 17:17.03 | Kobaz | well when the call is placed, the dialplan is hit to handle the call |
| 17:17.10 | leifmadsen | jaytee: no worries, because vinyl is coming back since it has better audio quality than CDs. No audio compression. |
| 17:17.15 | Kobaz | if the phone perhaps added a sip header saying it was the result of hitting the transfer button |
| 17:17.18 | Kobaz | that would be cool |
| 17:17.21 | nny_1 | leifmadsen: thanks again, that's what i needed |
| 17:17.25 | leifmadsen | Kobaz: so do that then |
| 17:17.26 | Kobaz | i haven't found anything like that on polycom phones yet |
| 17:17.30 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:17.44 | leifmadsen | Kobaz: but from the viewpoint of Asterisk, I'm sure it's just another INVITE with zero relation to a "transfer" |
| 17:17.48 | jaytee | I have J.Geils Bloodshot album in red translucent vinyl |
| 17:17.49 | Kobaz | yeah |
| 17:17.55 | Kobaz | leifmadsen: that's what i saw in the sip debug |
| 17:18.11 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
| 17:18.26 | coppice | the vinyl picture disks used to be kinda fun |
| 17:18.30 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 17:18.41 | [TK]D-Fender | jaytee: If it makes you feel better they have a HDTV add-on that introduces simulated fuzz & sweeping lines for that old-school feel ;) |
| 17:18.51 | jaytee | hahaha |
| 17:19.45 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 17:20.01 | jaytee | we'd never get away with a giant rolling paper like they put in Cheech and Chong's Big Bambu album nowadays. too much political constipatio.....er....correctness. |
| 17:21.46 | leifmadsen | jaytee: not in the US maybe.... |
| 17:21.53 | leifmadsen | jaytee: you could get away with it in Canada I'm sure :) |
| 17:21.58 | errr | jaytee: I bought a cd back a few years ago that came with blunt wrappers |
| 17:22.17 | coppice | james blunt wrappers? :-\ |
| 17:22.48 | jaytee | well, that's because Canada is an enlightened country. not blighted and blinded by near-sighted entrenched right wing morons. |
| 17:24.32 | coppice | CDs are too small to have interesting covers like we used to get with vinyl. album covers used to win art awards |
| 17:25.57 | jaytee | yeah, remember Santana's Abraxas? |
| 17:26.24 | coppice | or Jethro Tull's Thick as a Brick |
| 17:26.33 | jaytee | god, I loved that album |
| 17:26.58 | jaytee | I saw Jethro Tull perform live twice. Ian Anderson is a frigging genius |
| 17:27.07 | hesco | aye he is |
| 17:27.13 | luckyaba | Are some VOIP phones just not able to be used with Asterisk? |
| 17:27.15 | jaytee | "really don't mind if you sit this one out" |
| 17:27.23 | luckyaba | I have an Altigen here that I can't get registerd |
| 17:28.20 | kb3ien | the home brew is comming together... is there a way for Queue to fail if there are no agents? |
| 17:28.48 | hesco | jaytee: there are those who would suggest that the Harper Administration might perhaps be changing that about the nation to the North |
| 17:29.53 | [TK]D-Fender | [13:27]<luckyaba>Are some VOIP phones just not able to be used with Asterisk? <- Certainly |
| 17:30.31 | jaytee | kb3in, what do you mean by fail? pass the call on in the dialplan? |
| 17:32.29 | kb3ien | yes. rather than spend time parking the call. |
| 17:32.38 | kb3ien | or queueing the call. |
| 17:32.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:33.45 | luckyaba | Are there ways to load custom firmware to these proprietary phones? |
| 17:34.03 | [TK]D-Fender | kb3ien: there are queues.conf parms to kick people out if there aren't members to take the call |
| 17:34.10 | kb3ien | i want the call to fall through to the next priority in the dialplan. |
| 17:34.12 | kb3ien | ah. |
| 17:34.13 | jaytee | joinwhenempty=no |
| 17:34.16 | kb3ien | danke. |
| 17:34.17 | [TK]D-Fender | luckyaba: google it. Nobody talks about those here. |
| 17:34.52 | kb3ien | its a messy area. and some legal issues may or maynot surround it. depending where you live. |
| 17:35.05 | kb3ien | nobody talks about those /HERE/ |
| 17:35.08 | jaytee | I'd never heard of an Altigen phone before today but I don't get out much anymore. |
| 17:36.15 | [TK]D-Fender | jaytee: You've HAVE to not get out much to have heard of them ;) |
| 17:36.30 | [TK]D-Fender | jaytee: They're the kind of products closeted buyers get stuck with |
| 17:36.54 | [TK]D-Fender | jaytee: "Hi my boss liked their sales guy... am I screwed now?" |
| 17:37.08 | jaytee | [TK]D-Fender, by closeted are you trying to infer something? Cuz I own a sword too ya know :-) |
| 17:37.56 | outtolunc | just 1? |
| 17:38.21 | jaytee | well, I own more than one but what's the point of using 2 at the same time? |
| 17:38.29 | luckyaba | Well Cisco phones are in the same boat as Altigen I believe and I know for a fact those work with Asterisk |
| 17:38.51 | Qwell | same boat? do the Altigen phones suck too? |
| 17:38.51 | luckyaba | I was simply looking for info.... |
| 17:38.59 | luckyaba | because I have "googled it" |
| 17:38.59 | Qwell | I suspect they aren't even in the same ocean. |
| 17:39.06 | Qwell | (that was funnier in my head) |
| 17:39.17 | [TK]D-Fender | jaytee: double-entendres abound! |
| 17:39.20 | jaytee | so now we have 3 suck phones? not just Grandstream and Cisco any longer? |
| 17:39.30 | carrar | WHAT |
| 17:39.36 | carrar | Cisco phones LOOK sexy |
| 17:39.45 | jaytee | they do, I'll admit that |
| 17:39.45 | carrar | thats at least +1 |
| 17:39.50 | [TK]D-Fender | carrar: And you DO end end up feeling screwed ;) |
| 17:39.51 | luckyaba | haha |
| 17:39.56 | carrar | hahah |
| 17:40.00 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
| 17:40.13 | luckyaba | The nortel has a bluetooth ready phone that is pretty slick looking |
| 17:40.14 | carrar | only when I am looking for the BLF option |
| 17:40.17 | luckyaba | forget what model |
| 17:40.22 | jaytee | ooooh, shiny!!! owww, stop it that hurts!!! whadday mean, licensing? |
| 17:40.43 | carrar | .. and 50 other options |
| 17:40.59 | jaytee | luckyaba, don't leave your children unattended near a Nortel phone |
| 17:41.14 | luckyaba | hahaha? |
| 17:41.21 | carrar | and my 7941 does g722!! |
| 17:41.27 | carrar | heh |
| 17:41.31 | luckyaba | If I had children I might be inclined to ... what would happen? |
| 17:41.39 | jaytee | whoop-de-frikken-do, g722 |
| 17:41.44 | carrar | heh |
| 17:41.53 | coppice | jaytee: what happened to the consumer oriented governments of the 60s and 70s? how did things get so screwed up you have to agree to a bloody MS software licence to drive away your new car? |
| 17:41.57 | jaytee | luckyba, let's just say they'd need "therapy" |
| 17:42.43 | jaytee | coppice, yeah! I was wondering about that myself just the other day. I think Nader poisoned his own well running for President too many times. |
| 17:43.53 | coppice | jaytee: its not just the US, so its more than just nader |
| 17:45.22 | jaytee | coppice, well from what I understand there's an undisclosed bug in the software that after several months the system will respond to voice recognition requests for Michael Bolton or Neil Sedaka tunes only. |
| 17:45.31 | jaytee | damn! |
| 17:53.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:53.54 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 17:54.19 | VoipForces | Anyone looking for an asterisk/telehony job near Montreal? |
| 17:54.32 | timeshell_atwork | What kinda job? |
| 17:54.51 | [TK]D-Fender | Make me an offer :) |
| 17:55.05 | VoipForces | [TK]D-Fender: Trust me you are over qualified |
| 17:55.12 | [TK]D-Fender | VoipForces: lol |
| 17:55.25 | VoipForces | timeshell_atwork: Configuration, deployement and support |
| 17:55.29 | [TK]D-Fender | VoipForces: "no such thing as overkill" :p |
| 17:55.34 | timeshell_atwork | Oh, so it doesn't pay much then? |
| 17:56.10 | VoipForces | timeshell_atwork: I'm not in control of the pay, I'm not the boss. But must be willing to do 1st line support and customer site installs |
| 17:56.32 | VoipForces | [TK]D-Fender: Would you be willing to do first line support? |
| 17:56.34 | timeshell_atwork | Sounds like it doesn't pay much. |
| 17:57.28 | VoipForces | timeshell_atwork: Well, for sure it's not in the 100k cause I would be pissed-off LOL |
| 17:57.45 | timeshell_atwork | Well, if it's not 6 figure, I'm not interested. |
| 17:57.47 | timeshell_atwork | :D |
| 17:58.03 | [TK]D-Fender | VoipForces: Open for discussion. |
| 17:58.46 | VoipForces | [TK]D-Fender: What do you mean? |
| 17:59.46 | VoipForces | Real title would be more like Asterisk/Telephony field tech |
| 18:00.09 | *** join/#asterisk jsgoecke (n=Adium@c-71-202-25-141.hsd1.ca.comcast.net) |
| 18:01.15 | jaytee | damn, a choice position like that opens up and I don't have an updated passport! |
| 18:02.50 | VoipForces | jaytee: LOL our border is probably like a swiss cheese anyway LOL |
| 18:03.33 | jaytee | yeah, alsace lorraine. lots and lots of tiny little holes. kinda like the security in Windows |
| 18:03.44 | VoipForces | Anyway if anyone interested, contact me off channel |
| 18:03.53 | *** join/#asterisk Sheeplet (n=BuRn@216.32.93.241) |
| 18:04.18 | VoipForces | jaytee: Please don't compare Canada to Windows, that IS a real insult LOL |
| 18:04.59 | jaytee | I wasn't. I was comparing US border security to Windows security. The Canadian border guards are hard core hardasses |
| 18:05.14 | jaytee | they wouldn't let my friend Jack in because "he looked nervous" |
| 18:05.27 | jaytee | Jack always looks nervous |
| 18:05.55 | jaytee | plus they thought an RPG game on his computer was terrorist plans of some kind and wallpaper of Summer Glau was kiddie porn. |
| 18:06.44 | VoipForces | jaytee: LOL Last summer I was bringing my boy to a camp and made a wrong turn and endedup at the NY border. Had to explain the US border patrol that I just wanted to do a U turn... I had a brand new van I had picked up at the dealer the same day and had forgotten my papers. All I had was my driver liscence |
| 18:06.52 | VoipForces | I WAS nervoous LOL |
| 18:08.34 | jaytee | yeah, I tried to cross at Niagara when I was moving cross country and forgot I had my parakeets in the backseat. I almost wasn't allowed back into the US and Canada wasn't going to let me in so for awhile it was looking like I was going to have to live the rest of my life on the bridge. |
| 18:09.28 | Sheeplet | they have internet access at the bridge? |
| 18:09.33 | Sheeplet | wifihotspot or what/ |
| 18:09.34 | Sheeplet | ? |
| 18:09.46 | jaytee | not back then, don't know about now |
| 18:10.18 | jaytee | back then the internet was mostly text only and animated gif files. no sites had Flash or anything fancy. |
| 18:10.47 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 18:10.53 | jaytee | and public wifi was non-existent. hell, 802.11a or b didn't even exist in consumer devices yet. |
| 18:11.40 | nullable_type | Hey guys, I a doing third party calling to connect two phones using a voip provider. There is a big lag, is there any way i can optimize |
| 18:11.53 | nullable_type | I am using g729 codec |
| 18:11.57 | jaytee | I remember back when I thought 640x480 with 256 colors and a 2400baud modem made me a badass |
| 18:12.04 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 18:12.16 | timeshell_atwork | nullable_type : That is a very broad question |
| 18:12.35 | timeshell_atwork | The internet is only as fast as its slowest link. |
| 18:12.37 | jaytee | puts on Bruce Springsteen's "Glory Days" and waxes nostalgically. |
| 18:13.14 | timeshell_atwork | nullable_type : Start by having computers at both ends of the voip connection ping the provider. |
| 18:13.26 | timeshell_atwork | What are the ping times like? |
| 18:13.36 | nullable_type | timeshell >> we have our asterisk server at the voip provider's network to reduce internet travelling... the traceroute was excellent between the asterisk and the provider |
| 18:14.12 | VoipForces | nullable_type: the two phones are they using g729 also ? |
| 18:14.53 | timeshell_atwork | nullable_type : YOu can have latency from 1. Point A to provider 2. Point B to provider. 3. Codec translation 4. Poor bandwidth 5. Slow server |
| 18:14.56 | VoipForces | nullable_type: what about the traceroute between the phones and the asterisk server ? |
| 18:16.21 | nullable_type | VoipForces >> We are using the same Voip Provider who use g720 for both ends. And both phones are landline phones that the voip provider connects |
| 18:16.55 | nullable_type | timeshell >> If both legs use the same codec, we can cut down the codec translation rite |
| 18:17.32 | [TK]D-Fender | [14:11]<jaytee>I remember back when I thought 640x480 with 256 colors and a 2400baud modem made me a badass <- I've had almost a dozen computers far worse off than that |
| 18:17.57 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
| 18:18.15 | jaytee | [TK]D-Fender, yeah but back then it was "da bomb" |
| 18:19.25 | jaytee | [TK]D-Fender, course back then you were probably in third grade and hacking the power grid and the school system's computers with your Commodore Amiga :-) |
| 18:19.44 | [TK]D-Fender | jaytee: Amiga? Too new :p |
| 18:19.52 | [TK]D-Fender | jaytee: I had a PET & a VIC20 |
| 18:20.08 | [TK]D-Fender | jaytee: And several computers i couldn't even name with ROM BASIC, etc |
| 18:20.08 | jaytee | wow! |
| 18:20.18 | VoipForces | nullable_type: Is the asterisk server loaded? If not what is your internet pipe throghput? |
| 18:20.42 | hfx-ed | VIC20 rocked! |
| 18:21.06 | jaytee | [TK]D-Fender, I've actually used perforated paper tape. Can you beat that? |
| 18:21.42 | [TK]D-Fender | jaytee: I debug with a magnifying glass on a clear day :p |
| 18:21.50 | jaytee | hahahahaaa |
| 18:23.19 | timeshell_atwork | hfx-ed : Are you whacked? VIC20 was one of the worst home computers that ever existed. |
| 18:23.27 | outtolunc | punch cards will be all the rage (again) someday <G> |
| 18:23.50 | timeshell_atwork | Along with the Adam |
| 18:24.34 | hfx-ed | timeshell_atwork: I still remember the world of opportunity that the 8k memory expander opened up for us |
| 18:25.31 | hi365 | can someone calrify the difference between the tty optiond and the console option in safe_asterisk? |
| 18:26.02 | hfx-ed | timeshell_atwork: I admit though I did have a lot more fun using the C64 to control model trains and the Tandy Armatron |
| 18:29.03 | *** join/#asterisk cliff_ (n=cliffc@suid.net) |
| 18:29.34 | *** join/#asterisk cliff_ (n=cliffc@suid.net) |
| 18:31.54 | timeshell_atwork | hfx-ed : Vic20 would have had a saving grace had it had the same screen resolution of the C64 |
| 18:32.03 | timeshell_atwork | Alas, it wasn't to be. Vic20 officially sucked. |
| 18:32.58 | neurosys | [TK]D-Fender: IP Phone... Cisco or Polycom? |
| 18:33.11 | [TK]D-Fender | Polycom > All |
| 18:33.19 | neurosys | heh :) |
| 18:33.31 | *** join/#asterisk Greyer (n=Greyer@212.91.29.33) |
| 18:33.44 | timeshell_atwork | ? >= Polycom? |
| 18:33.51 | neurosys | NULL |
| 18:33.59 | timeshell_atwork | lol |
| 18:34.05 | timeshell_atwork | NULL >= Polycom |
| 18:34.11 | timeshell_atwork | That means nothing is just as good. |
| 18:34.13 | timeshell_atwork | :D |
| 18:34.13 | Greyer | hi, I'm looking for logwatch script to check asterisk logs, anyone wrote that? |
| 18:34.45 | [TK]D-Fender | polycom >= polycom |
| 18:34.57 | [TK]D-Fender | I'd put my money on the "equal" aprt of that |
| 18:35.04 | neurosys | heh |
| 18:35.25 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:36.17 | neurosys | if (neurosIpPhone != polycom) { exit(1) }; |
| 18:36.59 | neurosys | if (neurosIpPhone != polycom) { printf("you suck"); exit(1) }; |
| 18:37.02 | neurosys | :-D |
| 18:37.55 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 18:39.01 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 18:40.01 | *** join/#asterisk [Rated-R] (n=b0redum@bookit-dev.com) |
| 18:40.05 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 18:40.19 | *** part/#asterisk [Rated-R] (n=b0redum@bookit-dev.com) |
| 18:41.05 | seb- | [TK]D-Fender: hello..i just had a long irc chat w/ ekiga dev and he couldn't see anything wrong w/ my ekiga settings..is there ANY chance the problem is on your end? perhaps you have some firewall or NAT that prevents you from RECEIVING my audio even though you can SEND it? |
| 18:41.23 | *** part/#asterisk korihor (n=korihor@190.72.234.118) |
| 18:41.35 | seb- | [TK]D-Fender: i really should find some other ekiga user to test to rule that out |
| 18:42.30 | [TK]D-Fender | seb-: .... I'm the consultant, its my server, it is public and I wrote the bloody guide for configuring * & NAT :p |
| 18:43.17 | seb- | [TK]D-Fender: yea...i'm skeptical too...have you successfully done SIP chats with others from your home and hear their audio? (just checking) |
| 18:43.32 | [TK]D-Fender | seb-: You heard me in the confreence, I heard the prompts prior to entering. Audio both ways confirmed |
| 18:43.54 | seb- | [TK]D-Fender: true |
| 18:44.22 | seb- | [TK]D-Fender: it is weird that even ekiga devs can't see what wrong in my ekiga -d 4 output (verbose debugging output) |
| 18:45.16 | *** part/#asterisk Greyer (n=Greyer@212.91.29.33) |
| 18:45.20 | seb- | [TK]D-Fender: do you know any other consultants in SD? perhaps i can drive my laptop to them and.....who knows |
| 18:45.30 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 18:46.06 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
| 18:47.21 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-4b0243e0b472f70b) |
| 18:47.53 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:48.57 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-e7329eb346467ae8) |
| 18:50.44 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 18:50.48 | [TK]D-Fender | seb-: We'll try a more direct test together |
| 18:50.52 | [TK]D-Fender | (later) |
| 18:51.25 | seb- | [TK]D-Fender: ok..i set up an ekiga.net accout....that would rule out my * server |
| 18:51.30 | seb- | maybe that is an option |
| 18:52.16 | seb- | [TK]D-Fender: or whatever you think best...i'll await your instructions later..thanks |
| 18:52.28 | *** part/#asterisk acehunky (n=acehunky@123.252.144.92) |
| 18:53.47 | [TK]D-Fender | seb-: I'll set you up on MY server and we can test from there |
| 18:54.27 | timeshell_atwork | seb- Windows FW? |
| 18:54.38 | seb- | [TK]D-Fender: oh boy...my asterisk server's rep is in danger now! :) |
| 18:54.42 | seb- | timeshell_atwork: linux |
| 18:54.51 | timeshell_atwork | seb- Linux FW? |
| 18:54.53 | timeshell_atwork | :D |
| 18:55.13 | timeshell_atwork | I blame your linux config |
| 18:55.20 | seb- | timeshell_atwork: good guess |
| 18:56.18 | seb- | [TK]D-Fender: your not going to believe this but i think timeshell_atwork may be right |
| 18:56.36 | seb- | [TK]D-Fender: i setup my firewall to accept incoming audio ports but not let outgoing |
| 18:56.48 | seb- | [TK]D-Fender: later before we do your other test we should do a test w/ my fw turned off |
| 18:56.54 | seb- | [TK]D-Fender: then when it works i'll cry |
| 18:57.05 | seb- | [TK]D-Fender: of sadness or happiness i'm not sure |
| 18:57.50 | seb- | timeshell_atwork: you like an angel that appeared from nowhere |
| 18:58.34 | seb- | [TK]D-Fender: i think i'm crying already |
| 18:58.39 | seb- | :) |
| 18:58.49 | jaytee | I'm crying and it has nothing to do with any of this |
| 18:59.49 | jaytee | there's a pic of a woman in an advertisement for EarnMyDegree.com on the msnbc.com page and just knowing she exists and that I can't have her is making me cry. |
| 19:00.09 | seb- | jaytee: thwarted love...ah yes..that is crying material :) |
| 19:00.35 | seb- | jaytee: still fwiw...newbie voip hurdles would make the hardies man cry it seems |
| 19:00.42 | seb- | hardiest* |
| 19:00.50 | jaytee | who said anything about love? i just want to keep her in a pit in my basement :-) |
| 19:00.55 | neurosys | muhaha! |
| 19:01.33 | jaytee | and normally I don't go for women who wear bangs but she manages to pull it off nicely. |
| 19:01.49 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 19:04.52 | *** part/#asterisk VoipForces (n=kvirc@mail.viatransint.com) |
| 19:07.23 | *** join/#asterisk bertoskiz (n=chatzill@rrcs-67-78-186-27.se.biz.rr.com) |
| 19:10.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:12.35 | bertoskiz | if i wanted to limit outgoing calls via password for outbound trunks how would i pass the password to an IVR? |
| 19:14.03 | neurosys | bertoskiz, authenticate()? |
| 19:14.25 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.254) |
| 19:15.25 | bertoskiz | not sure..thats why im asking..lol |
| 19:15.26 | *** join/#asterisk twanny796 (n=chatzill@85.232.206.65) |
| 19:15.58 | twanny796 | cannot get the mic working with X-Lite on linux :( |
| 19:16.01 | bertoskiz | recently had a box hacked and someone scripting calls out asking for pins and such... |
| 19:16.05 | neurosys | bertoskiz, that would be my 1st guess. Check out the authenticate() function and see if its what you could use |
| 19:16.27 | bertoskiz | i will give it a shot..thanks |
| 19:16.46 | bertoskiz | now there has to be some way..just checking for suggestions |
| 19:17.43 | neurosys | bertoskiz: authenticate() will propmt for a password on the line. you pass the code as the paramater in your dialplan |
| 19:17.51 | *** part/#asterisk nny_1 (n=scott@64.203.244.146) |
| 19:18.13 | neurosys | bertoskiz: you can also pass it a standard text file and list multi codes on each line |
| 19:18.26 | neurosys | bertoskiz: I use this for pin #'s |
| 19:19.39 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:22.26 | bertoskiz | im sure that is just what i was looking for..thanks |
| 19:22.35 | bertoskiz | you rock : ) |
| 19:22.40 | twanny796 | cannot get the mic working with X-Lite on linux :( |
| 19:22.43 | neurosys | glad I could finally help someone ;) |
| 19:23.04 | nullable_type | Does anyone know how to diagonize the call latency issues |
| 19:23.13 | nullable_type | Other than SIP and RTP logging |
| 19:23.14 | bertoskiz | ill come back and shower you with praise..if i can get it to function correctly |
| 19:23.37 | bertoskiz | some providers will help with that |
| 19:23.51 | bertoskiz | latency stuff |
| 19:23.52 | infernix | Qwell: ping? |
| 19:24.05 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
| 19:24.11 | nullable_type | bertos >> Provider is not helping :( |
| 19:24.27 | bertoskiz | that could mean they suck..lol |
| 19:24.41 | bertoskiz | sorry thats not helpfull either |
| 19:24.52 | nullable_type | I know, They probably does, but i am told to work with them |
| 19:25.51 | nullable_type | "told" = since its recession it is easy for an employer to tell any stuff these days |
| 19:26.00 | bertoskiz | sorry i wish i could be of better service..you could try a packet sniffer..but wireshark is not the easiest thing to diagnose with |
| 19:26.24 | *** join/#asterisk kekoeoo (n=kekoeoo@213.249.63.18) |
| 19:26.26 | nullable_type | i tried rtp debug though, it seems it take a while to initialize rtp sessions but not sure why |
| 19:26.39 | kekoeoo | I am getting a 603 response instead of hangup in a SIP call |
| 19:26.48 | bertoskiz | but it will tell you everything thats happening in your network |
| 19:26.57 | kekoeoo | if I originate the call, and use soft hangup $channel, I get the hangup event |
| 19:27.20 | kekoeoo | if the other user hangs up though, I just get a 603 SIP response - what could be causing this? |
| 19:27.42 | kekoeoo | when I finally kill the channel, the cause isn NORMAL but CALL_REJECTED |
| 19:28.05 | kekoeoo | (it knows it was hungup after I hang it up on asterisk, but again, after other party has hung up) |
| 19:28.12 | nullable_type | Is g729 codec slower to use than others? may be that cause my call latency |
| 19:28.32 | kekoeoo | what can I check? Ive googled and seen a lot about 603, but when trying to originate a call |
| 19:28.34 | kekoeoo | nothing on hangups |
| 19:29.18 | *** join/#asterisk stijnbe (n=stijnbe@78-21-61-204.access.telenet.be) |
| 19:30.52 | kekoeoo | is there a way, using asterisk java, to observe more SIP events? or in dialplan catch them and set some variable |
| 19:32.39 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 19:33.48 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:35.58 | kekoeoo | no ideas on why I get 603 response instead of a hangup? |
| 19:36.27 | kekoeoo | or how I can listen to SIP events in dialplan and pass them over fastagi? |
| 19:39.39 | kekoeoo | Can I set a variable when I receive a SIP reponse in dialplan?? |
| 19:40.38 | nullable_type | Can you guys suggest a really good voip provider who can support Sip Reinvite? |
| 19:46.18 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
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| 19:54.49 | kekoeoo | perhaps the problem is just SIP hangup detection - what configuration can govern that? |
| 19:55.49 | *** part/#asterisk jsgoecke (n=Adium@c-71-202-25-141.hsd1.ca.comcast.net) |
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| 20:03.10 | nullable_type | Can you guys suggest a Qos Tool i can use to find call latency issues |
| 20:03.31 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 20:04.16 | kekoeoo | nullable_type, Ive not tried to debug latency, but try sip set debug on |
| 20:04.56 | kekoeoo | but, I guess this isnt going to show the problem... perhasp the MTU, buffering, jitter.... hrm, I am clueless |
| 20:05.09 | nullable_type | ya i tried that also rtp debug, but looking for anything better |
| 20:05.31 | nullable_type | i am wondering if its the g729 codec, my voip provider only supports it |
| 20:06.25 | _ShrikE | nullable_type: capture your media traffic and laod it into wireshark |
| 20:06.32 | _ShrikE | err.. load |
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| 20:15.09 | *** join/#asterisk frek310 (n=herman@72.37.252.50) |
| 20:15.14 | frek310 | hello |
| 20:16.05 | frek310 | I'm using asterisk 1.2 and need my sip trunks to re-register every 60 seconds. Is there a setting that I can set for this? |
| 20:16.38 | nullable_type | Shrike >> thanks |
| 20:17.17 | nullable_type | Do you guys know if "183 Session Progress" is understood by asterisk, it seems like it was trying to bridge as soon as it get that, instead of waiting for 200 OK |
| 20:25.47 | nullable_type | is PCMU = ulaw? I enabled ulaw in sip.conf but i only see pcmu in sip logs |
| 20:29.49 | Kobaz | pcmu is pcmu... ulaw us ulaw |
| 20:30.21 | nullable_type | are they not same |
| 20:30.40 | gr0mit | they are the same |
| 20:32.39 | nullable_type | thank you |
| 20:35.27 | wwalker | anyone using AMD and having good detection rates? If so, what time arguments are you using? |
| 20:36.26 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-afb1256aa3742231) |
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| 20:45.55 | _ShrikE | nullable_type: 183 is early meida. Hence the bridging. |
| 20:50.24 | *** part/#asterisk icyValk77 (n=icyValk7@host86-161-124-210.range86-161.btcentralplus.com) |
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| 20:55.02 | nullable_type | _Shreike, is it supposed to start the bridging or to wait till 200 |
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| 21:04.16 | nullable_type | Do you guys have a good suggestion for a VOIP provider |
| 21:05.46 | mmlj4 | teliax |
| 21:07.02 | beek | nullable_type: vitelity is good, as well. I use both. |
| 21:07.40 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:07.41 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:08.49 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 21:09.27 | nny_1 | if i symlink a voicemail folder into the root of another voicemail folder, does asterisk check recursivly in the INBOX for "new messages" |
| 21:12.04 | VaGoNeTaS | sombody knows if there is posible to deactivate the "DND Status" on Asterisk? |
| 21:12.37 | *** join/#asterisk diatonic1 (n=chillman@mail.clearwater-research.com) |
| 21:12.54 | [TK]D-Fender | VaGoNeTaS: WHAT DND? |
| 21:13.05 | diatonic1 | Do not Disturb? |
| 21:13.31 | VaGoNeTaS | do not disturb status |
| 21:13.45 | [TK]D-Fender | VaGoNeTaS: Exactly... WHAT DND? Where do you see this? |
| 21:13.52 | edibrac | are mechanical crackling sounds a result of not having proper echo cancellation? |
| 21:14.04 | VaGoNeTaS | on the CLI |
| 21:14.14 | [TK]D-Fender | edibrac: can contribute |
| 21:14.15 | VaGoNeTaS | i can see that some agents are cheating |
| 21:14.19 | edibrac | it's odd - we've been running off a sangoma A102 for 5 months and it's been fine |
| 21:14.36 | [TK]D-Fender | VaGoNeTaS: DND is in in *, its on the PHONE |
| 21:14.52 | VaGoNeTaS | they put themselves as DND on his softphones |
| 21:14.55 | VaGoNeTaS | like Xlite |
| 21:15.02 | VaGoNeTaS | i'd like to disable that option on * |
| 21:15.03 | [TK]D-Fender | yes, and there's nothing * can do about it |
| 21:15.10 | VaGoNeTaS | fuck |
| 21:15.15 | VaGoNeTaS | nothing at all? |
| 21:15.21 | VaGoNeTaS | what do u recomend me |
| 21:15.25 | VaGoNeTaS | changing the softphones? |
| 21:15.40 | Khratos | Ask for a custom X-Lite :/ |
| 21:15.41 | [TK]D-Fender | VaGoNeTaS: I recommend never using softphones period |
| 21:16.40 | [TK]D-Fender | Khratos: not really happening |
| 21:16.56 | VaGoNeTaS | so what should they use instead? |
| 21:16.58 | nny_1 | nm testing shows asterisk only checks the root of INBOX. I have a client wanting a general company mailbox, any way to associate the messages left in it with multiple user's voicemail? I can just give them a way to dial to it, just curious if I can do it |
| 21:17.11 | [TK]D-Fender | VaGoNeTaS: a decent phone you can control |
| 21:17.20 | Khratos | Oh well, I have only being able to disable that button on a polycom phone |
| 21:18.15 | [TK]D-Fender | nny_1: what does it have to do with multiple users? |
| 21:18.57 | nny_1 | [TK]D-Fender: er, i have a voicemailbox (call it 100) and if anyone leaves a message there, i want it to show up in 101 and 102's vm INBOX |
| 21:19.23 | [TK]D-Fender | nny_1: How would they leave it there? |
| 21:19.30 | nny_1 | through the dialplan |
| 21:19.39 | nny_1 | Voicemail(100@default,u) |
| 21:19.43 | kaldemar | core show application voicemail |
| 21:19.44 | [TK]D-Fender | nny_1: then call Voicemail passing it multiple boxes |
| 21:19.49 | [TK]D-Fender | ^^ |
| 21:19.52 | nny_1 | k |
| 21:20.01 | nny_1 | that's cool, unaware of that |
| 21:20.04 | edibrac | can anyone here recommend a consultant that has experience troubleshooting low level problems with asterisk (line noise; PRI-level debugging)? |
| 21:20.18 | edibrac | ..what can I expect to pay for that? |
| 21:20.34 | VaGoNeTaS | [TK]D-Fender a decent phone u mean a SIP Hardware Telephone' |
| 21:20.36 | VaGoNeTaS | ? |
| 21:20.43 | [TK]D-Fender | VaGoNeTaS: YES... |
| 21:22.09 | VaGoNeTaS | got it |
| 21:25.49 | VaGoNeTaS | and |
| 21:25.58 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 21:26.02 | VaGoNeTaS | it is possible to trace an agent when he is putting himself as DND? |
| 21:26.20 | VaGoNeTaS | with Avaya is possible to trace an agent |
| 21:28.12 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
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| 21:40.08 | [TK]D-Fender | VaGoNeTaS: You'd have to monitor SIP debug and see the reject code and associate that to a call that should be be rejected. |
| 21:53.46 | seb- | [TK]D-Fender: tell me if and when you have time to test |
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| 22:11.34 | wwalker | is there any way to get time stamps on the messages in the asterisk console? |
| 22:14.43 | *** join/#asterisk [gquit]bombadil (n=dana@cpe-174-102-203-145.wi.res.rr.com) |
| 22:15.25 | edibrac | i got a recording of our weird mechanical noises ... anyone care to guess what it is? |
| 22:15.29 | edibrac | http://davaconsulting.com/idicto_files/xk4wot4pmg8d04ybsb4hu0ypg/Record%200011%202009-06-02%2014-34-42.aiff |
| 22:15.55 | edibrac | or i wonder if anyone else here has heard it before |
| 22:16.15 | edibrac | and perhaps point me in the right direction. |
| 22:19.25 | *** join/#asterisk jtodd (i=tv1p2854@ns.fox-den.com) |
| 22:19.25 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 22:26.59 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 22:27.45 | rue_mohr | when a sip phone is using rfc2833 for dtmf mode, what determines the length of the tones created out the fxo card? |
| 22:28.32 | Kobaz | rue_mohr: your dahdi settings |
| 22:28.39 | rue_mohr | ah |
| 22:28.53 | rue_mohr | chan_dahdi? |
| 22:29.15 | russellb | that's not actually true |
| 22:29.36 | russellb | Assuming Asterisk 1.4 or later, we preserve the length of the incoming tone out the FXO |
| 22:29.39 | rue_mohr | does the card or asterisk generate the tone? |
| 22:29.49 | russellb | Asterisk (DAHDI, actually) |
| 22:29.55 | rue_mohr | oh, so its the phone |
| 22:30.02 | russellb | but Asterisk controls the length |
| 22:30.08 | rue_mohr | .... ok |
| 22:30.10 | russellb | and it controls the length by passing through the length that was received |
| 22:30.24 | rue_mohr | circle: |
| 22:30.29 | russellb | square: |
| 22:30.49 | rue_mohr | your right, there are a lot of corners |
| 22:30.53 | Qwell | pineapple" |
| 22:30.56 | Qwell | d'oh |
| 22:31.01 | russellb | Qwell: that's not a shape, you fail |
| 22:31.01 | rue_mohr | ok SO, I change the dtmf time on the sip phone |
| 22:31.53 | Qwell | russellb: how else would you describe the shape of a pineapple? |
| 22:31.53 | rue_mohr | edits the phone config |
| 22:31.53 | Qwell | HMM? |
| 22:31.53 | russellb | In theory, your SIP phone should be sending length based on how long you hold down the button |
| 22:31.53 | russellb | but maybe some phones have it set statically ... |
| 22:31.53 | rue_mohr | no |
| 22:31.53 | rue_mohr | its comming out a set time |
| 22:31.53 | russellb | lame! |
| 22:31.53 | russellb | :-) |
| 22:31.53 | rue_mohr | verry short too |
| 22:32.07 | russellb | are you using any DTMF controlled features? |
| 22:32.22 | russellb | because if so, that complicates things a bit. I think in that case, we don't actually preserve the incoming length right now ... |
| 22:32.27 | *** join/#asterisk ReD-MaN (i=rox-ur-s@209.183.147.106) |
| 22:32.28 | rue_mohr | well, dialing out our co line, I'd call that a feature |
| 22:32.33 | russellb | lol |
| 22:32.37 | rue_mohr | :) |
| 22:32.39 | russellb | I mean like '#' transfer or something |
| 22:32.49 | rue_mohr | among other featrues, like rining and call display ;) |
| 22:32.54 | russellb | right right\ |
| 22:33.01 | thehar | HELLOOS |
| 22:33.05 | russellb | features.conf stuff |
| 22:33.09 | russellb | OMG thehar ! |
| 22:33.13 | thehar | zomg |
| 22:33.18 | rue_mohr | yes, I'm TRYING to get *0 to wink the dahdi channel |
| 22:33.31 | thehar | lol that sounds incredibly dirty |
| 22:33.37 | russellb | well then. If so ... Asterisk is probably regenerating the DTMF at it's own specified length. |
| 22:33.44 | rue_mohr | but I'v temporarily put that aside as the receptionist seems to think this dialing thing is important |
| 22:34.14 | rue_mohr | I'm gonna look into the sip.conf, I'm SURE SOMEWHERE I have seen a dtmf time setting |
| 22:34.56 | rue_mohr | <PROTECTED> |
| 22:34.56 | rue_mohr | ^^^aha |
| 22:34.57 | rue_mohr | bets thats not 50 seconds |
| 22:35.17 | russellb | 50 ms, presumably. |
| 22:35.26 | rue_mohr | which all in all is pretty short |
| 22:35.41 | russellb | If you change that and it doesn't help ... edit this line in channel.c: |
| 22:35.47 | russellb | #define AST_DEFAULT_EMULATE_DTMF_DURATION 100 |
| 22:35.47 | rue_mohr | :) |
| 22:35.50 | rue_mohr | k |
| 22:35.52 | russellb | change 100 to whatever you want |
| 22:35.57 | russellb | and see if that improves things for you. |
| 22:36.05 | rue_mohr | CHANGE it to 100? |
| 22:36.09 | russellb | it is 100 now |
| 22:36.15 | russellb | by the way, what problem are you having? |
| 22:36.26 | russellb | most FXO dialing problems are just that you need to put a wait before the number. |
| 22:36.40 | russellb | Dial(DAHDI/1/www${NUMBER_TO_DIAL}) |
| 22:36.48 | rue_mohr | sometimes digits dont get picked up |
| 22:36.59 | russellb | do you know which digits? |
| 22:37.13 | rue_mohr | yes and no, our speed dials dial the dahdi channels, users wait for a tone, and go from there |
| 22:37.16 | russellb | there is a good chance it's the first one (or few), so try adding a wait |
| 22:37.24 | russellb | ah. |
| 22:37.24 | rue_mohr | ah, for echo to settle |
| 22:37.33 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-e7329eb346467ae8) [NETSPLIT VICTIM] |
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| 22:37.42 | rue_mohr | k, I'll catch up here |
| 22:37.42 | russellb | but anyway ... that's where the DTMF knobs are, heh |
| 22:37.44 | rue_mohr | changing the sip config is a start.. |
| 22:37.51 | jaytee | mine works great for digits 0 through 9 but lebenty-leven keeps dropping everytime :-) |
| 22:37.51 | rue_mohr | thankyou |
| 22:37.54 | russellb | you're welcome |
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| 22:39.25 | *** join/#asterisk ctooley (n=ctooley@76.22.158.92) |
| 22:39.53 | ctooley | How do I tell a sip peer to spin up the RTP earlier (ie: early media) |
| 22:40.29 | jaytee | spin up? sounds like someone's confusing RTP with an FTL drive on Galactica :-) |
| 22:40.41 | russellb | ctooley: hrm ... Progress() in the dialplan? |
| 22:41.06 | ctooley | russellb, this is for calls originated via the Manager interface. |
| 22:41.17 | ctooley | We're missing audio due to stupidity on the remote end |
| 22:42.40 | thehar | russellb: i'm probably coming to astricon this year =) |
| 22:42.58 | russellb | w00t |
| 22:43.03 | russellb | runs off ... softball time |
| 22:43.07 | thehar | byeee |
| 22:43.11 | jaytee | hit a homer! |
| 22:43.25 | Qwell | jaytee: their pitcher is named Homer! |
| 22:45.12 | jaytee | Qwell, is his last name Hickam and is into rocketry in a big way? |
| 22:46.20 | rue_mohr | ok, its not sip.conf |
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| 22:47.11 | infernix | Qwell: hey, have you tried G1 cupcake with chan_mobile yet? |
| 22:47.52 | infernix | still only noise on my end so just looking if someone has had success |
| 22:48.04 | edibrac | is this the sound of someone's iphone interfering with the call: |
| 22:48.06 | edibrac | http://davaconsulting.com/idicto_files/xk4wot4pmg8d04ybsb4hu0ypg/Record%200011%202009-06-02%2014-34-42.aiff |
| 22:48.33 | edibrac | i'm guessing that's one possibility - but then again, i've heard reports of this sound a few times today |
| 22:49.58 | rue_mohr | if there a realtime or .conf override for AST_DEFAULT_EMULATE_DTMF_DURATION somehwere, there must be |
| 22:52.39 | s14ck | is away: Estoy ocupado |
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| 23:07.04 | beek | evening jaytee |
| 23:07.15 | jaytee | evening beek |
| 23:08.49 | rue_mohr | russellb, interestingly enough, the zaptel flash that I have programmed in features.conf works better now with the rfc setting |
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| 23:29.53 | *** join/#asterisk EGBlue (n=EGBlue@rrcs-24-43-24-35.west.biz.rr.com) |
| 23:30.41 | EGBlue | Hey guys, I've got a call that I would like to transfer to a different sip and once the call has been answered, drop it on my side. right now I am using Dial to forward the call which is successful, but I want it to be released once it has been answered, can it be done? Thanks! |
| 23:31.07 | rue_mohr | ok in sip.conf, tone.dtmf.onTime is the amount of time it makes the tone in the earpeice |
| 23:31.36 | rue_mohr | EGBlue, why dont you just use transfer? |
| 23:32.02 | EGBlue | rue_mohr, I tried to use transfer, but for some reason when I do it does transfer the call, but there is no audio. |
| 23:32.12 | rue_mohr | ah, rtp problem |
| 23:32.25 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 23:32.26 | rue_mohr | have a firewall in there? |
| 23:32.32 | EGBlue | yep |
| 23:32.38 | EGBlue | does it use a different port? |
| 23:32.42 | rue_mohr | how many rtp slots do you have? |
| 23:32.46 | EGBlue | right now we allow only 5060 |
| 23:32.53 | rue_mohr | "Really!? That Port!? |
| 23:32.55 | rue_mohr | " |
| 23:33.01 | rue_mohr | no 10000+ |
| 23:33.05 | rue_mohr | carries the audio |
| 23:33.19 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 23:33.24 | rue_mohr | look up /etc/asterisk/rtp.conf |
| 23:33.40 | EGBlue | 1 sec ;) |
| 23:34.02 | EGBlue | rtpstart=10000 |
| 23:34.03 | EGBlue | rtpend=20000 |
| 23:34.13 | EGBlue | it means we have to open all the ports from 10000 to 20000? |
| 23:35.03 | rue_mohr | no, |
| 23:35.11 | rue_mohr | only the number of them you want audio for |
| 23:35.42 | EGBlue | so I need to open a few in the range between 10000 and 20000? |
| 23:36.26 | rue_mohr | yes |
| 23:36.39 | rue_mohr | start at 10000, it will go up as the calls stack up |
| 23:37.00 | rue_mohr | "but I dont know anything about how asterisk works" |
| 23:37.03 | EGBlue | gotcha, thank you very much for your help rue_mohr, I will try it out |
| 23:37.09 | EGBlue | ;) |
| 23:37.10 | rue_mohr | have fun! |
| 23:38.09 | rue_mohr | wonders why people never set up the rtp and forget the sip |
| 23:38.33 | rue_mohr | "I can hear the person I'm calling, but I cant call them" .... yea... ok.... |
| 23:42.20 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:45.49 | *** join/#asterisk dshap (n=dshap@ip70-181-91-110.oc.oc.cox.net) |
| 23:46.01 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 23:46.33 | dshap | is anyone here familiar with the SendDTMF() application? If I place an outgoing call from my server to my cell phone and have it wait a few seconds and then SendDTMF, shouldn't i be able to hear the tones? |