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| 02:16.06 | Redimido | Hi guys |
| 02:16.48 | Redimido | some one can help me to configure my speakers voicing? the voice comes out like a robot, I cannot understand it |
| 02:17.46 | Redimido | and I am tired of googling and finding nothing but complains about the voicing using the sound cards |
| 02:22.19 | Redimido | I mean, I configured correctly all ip phones voicing, but we have an area of the office where there are no phones, we want to use a speaker to notify the guys someone is looking for them |
| 02:22.52 | Redimido | so I set up the /dev/dsp and it works, but the quality is really bad and I have not succeeded on cfixing it |
| 02:26.46 | Redimido | is it doable to use the sound card to do the voicing? Sorry but I do not see any activity on the channel, I mean no disrespect, but really need a hand |
| 02:27.59 | Redimido | help servchan |
| 02:28.39 | dpilon | ask in #freepbx there might be people there |
| 02:28.58 | Redimido | ok thanks dpilon |
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| 02:49.45 | Redimido | I get no answer on #freepbx channel either |
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| 02:55.30 | carrar | get a SIP based PA |
| 02:55.51 | carrar | http://www.cyberdata.net/products/voip/digitalanalog/pagingampv2/index.html |
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| 02:57.34 | carrar | Redimido, you can use the sound card to drive a PA but it kinda lame |
| 03:01.02 | Redimido | thanks carrar |
| 03:01.07 | Redimido | what else can I do |
| 03:01.15 | Redimido | how about using a softphone with auto-answer |
| 03:01.20 | Redimido | or some voip phone to send the voice to the speakers |
| 03:01.38 | Redimido | I am checking and Twinkle has auto-answer and a command-line interfase |
| 03:02.00 | Redimido | just I have not configured any softphone like that before :-) |
| 03:08.37 | *** join/#asterisk baronobeefdip (~baronobee@216-82-199-195.dyn.grandenetworks.net) |
| 03:08.52 | baronobeefdip | hello, I need some help with my dialplans in my asterisk pbx server |
| 03:09.00 | AkkerKid | heya everyone! I imaged my machine to clone it and ended up having to reinstall grub and now my call quality is choppy... What could that be about? |
| 03:09.11 | Redimido | says "BRB" |
| 03:10.11 | baronobeefdip | I have prepared my configuration files for this request for assistance |
| 03:10.27 | AkkerKid | what's the problem beef? |
| 03:10.35 | baronobeefdip | for some reason when I dial an extension number my ip phone says there was a 404 error and that the user wasn't found |
| 03:10.56 | AkkerKid | sounds like your phone is not provisioned properly |
| 03:11.06 | baronobeefdip | i have clearly put in some sip statements and dialplans (which is what I am worried about) and it's still not working |
| 03:11.23 | AkkerKid | the phone may not even be connected to the asterisk box properly. |
| 03:11.41 | AkkerKid | do you see the phone connected when you do asterisk-rx "sip show peers" |
| 03:12.15 | baronobeefdip | it's actually my tablet runnign voiper, I know it's a proper softphone because i tried it with a trixbox server and it worked but I want to do this on a machine that I already have debian installed on along with some other stuff and I don't have the equipment to create a dedicated asterisk box |
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| 03:12.35 | baronobeefdip | I am properly connected to the pbx box, or else it wouldn't have registered itself to the server, |
| 03:12.46 | baronobeefdip | I put in the ip address and port number 5060 |
| 03:12.52 | AkkerKid | got it. |
| 03:13.22 | baronobeefdip | still nothing, I have prepared something on pastebin for review (the extension.conf file is what I think is the problem but I will post my sip.conf file too) |
| 03:13.32 | AkkerKid | and you are handing calls from this extention to a context that list your destination as one of the options? |
| 03:13.40 | AkkerKid | link me please |
| 03:14.52 | baronobeefdip | http://pastebin.com/q7xA2hMC |
| 03:15.21 | baronobeefdip | I am only calling a softphone to a softphone for experimental purposes |
| 03:15.38 | baronobeefdip | when I feel ready I will move up to the more advanced stuff but this is pretty basic for me |
| 03:16.34 | AkkerKid | nat=yex? |
| 03:16.39 | AkkerKid | yes*? |
| 03:17.35 | baronobeefdip | I am not using nat but sure. |
| 03:17.55 | AkkerKid | do you have anything in your sip_nat.conf file? |
| 03:17.57 | baronobeefdip | nat is at yes |
| 03:18.13 | AkkerKid | i doubt there is nat between your phone system and your extensions... |
| 03:18.37 | baronobeefdip | I am not using nat right now, everything is happening within the LAN, I'll step iutside of the LAN a little later |
| 03:19.20 | AkkerKid | can you initiate a call from the CLI to one of your devices? |
| 03:19.21 | baronobeefdip | I think the problem is in the extensions.conf file since I can register a SIP device fine I assume everything in the sip.conf file are set correctly but I will let you be the judge of that |
| 03:20.27 | baronobeefdip | idk how to do that, even if I did I don't think it will work at this point since I can't even call the devices with the softphones and I know that that softphones are configured correctly because they successfully registered with the server they just don't call and talk to each other like I want them to |
| 03:20.48 | baronobeefdip | what changes to I need to make to the extensions.conf file and what should be changed in the sip.conf file if anything |
| 03:21.17 | AkkerKid | i just want to confirm that the receiving device can get a call... |
| 03:21.24 | AkkerKid | that would confirm half of your dialplan |
| 03:21.26 | baronobeefdip | it can't |
| 03:21.45 | baronobeefdip | I tried callng them numeriously and they don't call or ring at all |
| 03:22.05 | AkkerKid | but i'm saying, have you tried calling them from the asterisk box? |
| 03:22.12 | baronobeefdip | I only have the Dial command in the dial plans for both devices in my extensions.conf file |
| 03:22.14 | AkkerKid | not from an extension |
| 03:22.15 | baronobeefdip | yes |
| 03:22.28 | baronobeefdip | the softphone is on the asterisk box while the other is on a tablet |
| 03:23.19 | baronobeefdip | to answer a questions you asked earlier, I don't have a sip_nat.conf file |
| 03:23.29 | AkkerKid | understood |
| 03:23.31 | baronobeefdip | What needs to be done to the dialplan |
| 03:23.38 | baronobeefdip | that I currently have |
| 03:24.11 | AkkerKid | what happens if you run this: channel originate SIP/201 extension 203@home |
| 03:24.20 | AkkerKid | run that from CLI and tell me what happens |
| 03:26.35 | AkkerKid | anything? |
| 03:26.54 | baronobeefdip | the phone rings on the 201 device |
| 03:27.17 | AkkerKid | and when you pick up? |
| 03:28.05 | baronobeefdip | I get a congratulatory message |
| 03:28.09 | baronobeefdip | for installing asterisk |
| 03:28.33 | AkkerKid | it should have rung your 203 extension |
| 03:28.41 | baronobeefdip | it did |
| 03:28.46 | baronobeefdip | not |
| 03:29.11 | baronobeefdip | I put in the command and i get a congratulatory message with a female voice asking me to dial some numbers for some testing processes |
| 03:29.14 | AkkerKid | and when you edit this file you reload it to asterisk each time right? |
| 03:29.29 | baronobeefdip | the extensions.conf file? |
| 03:29.32 | AkkerKid | yeah |
| 03:29.41 | baronobeefdip | if so then yes, every time I add something to it i use this command |
| 03:29.47 | baronobeefdip | /etc/init.d/asterisk restart |
| 03:29.57 | AkkerKid | that's one way to do it... |
| 03:30.06 | baronobeefdip | still doesn't make it work though |
| 03:30.20 | AkkerKid | you could do this too. asterisk -rx "reload" |
| 03:30.23 | AkkerKid | but anyway |
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| 03:31.21 | baronobeefdip | the caller if on the call says asterisk if that gives any clues |
| 03:31.35 | AkkerKid | weird... |
| 03:31.50 | baronobeefdip | So now I know that I can call a device though the CLI but what do i put into the extentions.conf file for a workable and basic dial plan |
| 03:32.41 | baronobeefdip | because it still isn't working and I only have the Dial command for each extension and nothing else and I am thinking that that may be one of the problems but you are the expert |
| 03:32.54 | AkkerKid | i wishi i was an expert |
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| 03:33.06 | baronobeefdip | do you know what I should put into the extensions.conf file |
| 03:33.17 | baronobeefdip | thats pretty much all I need to know at this point |
| 03:33.39 | AkkerKid | honestly, what you have looks fine |
| 03:34.19 | baronobeefdip | but it doesn't work |
| 03:34.39 | baronobeefdip | i can't call anything because of a 404 error |
| 03:35.54 | AkkerKid | can you watch the CLI as you make the call? |
| 03:36.04 | AkkerKid | paste what happens in the CLI |
| 03:36.07 | baronobeefdip | I did and I get some kind of error |
| 03:36.23 | baronobeefdip | [Jan 3 22:25:42] WARNING[5268]: chan_sip.c:3914 retrans_pkt: Maximum retries exceeded on transmission 4ccd5882-8c54-e211-99ed-0013f7e94671@debian for seqno 5 (Critical Response) -- See doc/sip-retransmit.txt. |
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| 03:36.27 | baronobeefdip | theres the error |
| 03:36.31 | baronobeefdip | I have no idea what it means |
| 03:37.06 | baronobeefdip | It might be the problem though |
| 03:37.26 | AkkerKid | may even be a firewall issue |
| 03:37.47 | baronobeefdip | firewall chains are clear and allowing everything |
| 03:37.55 | baronobeefdip | heres the other error that came on sometime later |
| 03:37.56 | baronobeefdip | [Jan 3 22:25:22] NOTICE[5268]: chan_sip.c:21533 handle_request_subscribe: Failed to authenticate device <sip:203@192.168.1.44>;tag=34e05882-8c54-e211-99ed-0013f7e94671 for SUBSCRIBE |
| 03:38.26 | AkkerKid | ok so extension 203 isn't properly conected |
| 03:39.11 | baronobeefdip | device 203 is the softphone and it should be since it is registered and I put in all of the information in the ekiga client correctly just like I have been doing with the trixbox setups |
| 03:39.32 | AkkerKid | ekiga or voiper? |
| 03:39.36 | AkkerKid | one of each? |
| 03:39.51 | baronobeefdip | ekiga is 203, the other is voiper |
| 03:40.02 | baronobeefdip | both registered to the server successfully |
| 03:40.31 | baronobeefdip | but won't talk to each other |
| 03:41.23 | baronobeefdip | I have made sure not to register both extensions on one of the clients. each one should have one extension registered to it and they are both different |
| 03:42.07 | AkkerKid | try running this: channel originate SIP/203 extension 201@home |
| 03:42.24 | AkkerKid | maybe we'll get a different outcome going the other direction? |
| 03:43.44 | baronobeefdip | no errors but I get the test lady again |
| 03:44.22 | baronobeefdip | still can't call the other devices though |
| 03:44.29 | AkkerKid | it almost sounds like something is hyjacking your extensions.conf file and replaceing it with the default |
| 03:45.14 | AkkerKid | the test lady is in the default extensions.conf file |
| 03:45.42 | AkkerKid | and if you've replaced that with your code, it isn't being run... |
| 03:45.43 | baronobeefdip | what do you think I should put in the extensions.conf file then? from what you have seen at pastebin |
| 03:45.54 | baronobeefdip | because I have absolutely no clue |
| 03:46.35 | AkkerKid | maybe asterisk isn't loading your extensions.conf file. |
| 03:46.46 | AkkerKid | what are the permissions and user of that file? |
| 03:47.23 | baronobeefdip | i did a chown for asterisk to the extensions.conf file but what would you suggest for changing the permissions to the file if that is the case |
| 03:47.36 | AkkerKid | make sure it's 664 and asterisk:asterisk |
| 03:47.41 | AkkerKid | then reload asterisk |
| 03:47.49 | baronobeefdip | can you give me the command verbatim please |
| 03:47.56 | baronobeefdip | I am documenting everything as you tell me |
| 03:48.09 | baronobeefdip | because I don't want to go thorugh all this madness again |
| 03:48.09 | AkkerKid | chmod 664 /etc/asterisk/extensions.conf |
| 03:48.30 | AkkerKid | chown asterisk:asterisk /etc/asterisk/extensions.conf |
| 03:48.39 | AkkerKid | asterisk -rx "reload |
| 03:48.45 | AkkerKid | asterisk -rx "reload" |
| 03:49.15 | baronobeefdip | okay I just did the chown and chmod |
| 03:50.09 | AkkerKid | any difference after reload? |
| 03:52.26 | baronobeefdip | try it from the tablet to call 203 and I get an error again, I did it from the ekiga client and I don't get the user not found error but the tablet isn't ringing, I think I need to add stuff to the extensions. the only difference I have noticed is that ekiga now registers a voice mail box which it didn't do eariler |
| 03:53.08 | baronobeefdip | now that asterisk allegedly has ownership of the extensions.conf file, what needs to be done to the extensions.conf file now |
| 03:53.10 | AkkerKid | well, sorry but i have to get some sleep |
| 03:53.16 | baronobeefdip | okay |
| 03:53.27 | baronobeefdip | everyone keeps telling me to use trixbox |
| 03:53.47 | baronobeefdip | but I want to do it all from the text editor since gui inbterfaces tend to be very confusing |
| 03:53.52 | AkkerKid | i would do elastix before trixbox simply for trix's security flaws |
| 03:53.53 | baronobeefdip | and text is more precise |
| 03:54.21 | baronobeefdip | it doesn't matter what you use as a GUI i just want my text editing method to work |
| 03:54.29 | AkkerKid | check back here in 12 hours and more people would be able to help you |
| 03:54.35 | baronobeefdip | okay |
| 03:54.35 | AkkerKid | i'm out |
| 03:54.41 | AkkerKid | good luck |
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| 09:58.54 | bombev | Happy new year to all, I wish all the best :) |
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| 12:23.34 | bombev | which one is better paid g729 or the free g729 |
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| 12:28.09 | gavimobile | im trying to change my mysql cdr records to a remote location. editing the file "File: /etc/odbc.ini" and File: /etc/asterisk/cdr_mysql.conf to match with my remote mysql server config along with running "cdr show status" from cli shows this http://pastebin.com/YGPA8QW9 I didn't test yet, however I cannot do "core restart now" |
| 12:28.24 | gavimobile | it says No such command 'core restart now' (type 'core show help core restart now' for other possible commands) |
| 12:28.36 | gavimobile | it was working before I made changes. ill post my 2 files which I mentioned above |
| 12:30.27 | gavimobile | http://pastebin.com/nyCVUJxF && http://pastebin.com/2k3vVpbY |
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| 12:35.03 | gavimobile | restarted mysql, then restarted asterisk and now asterisk won't start |
| 12:35.28 | gavimobile | im looking in the logs and it says [2013-01-04 14:34:15] WARNING[5646] pbx.c: PBX requires Asterisk to be fully booted [2013-01-04 14:34:15] WARNING[5646] chan_sip.c: Failed to start PBX :( |
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| 14:38.33 | BorjaGVO | Hi people! Going again over "The Definitive Guide", I have a doubt on what these words exactly mean: Peer --> "Match incoming requests to a configuration entry using the source IP address and port number". |
| 14:38.34 | BorjaGVO | <PROTECTED> |
| 14:39.24 | BorjaGVO | for user--> "Match incoming requests to a configuration entry using the username in the From header of the SIP request. This name is matched to a section in sip.conf with the same name in square brackets." |
| 14:39.39 | kaldemar | yes, it means type=peer in sip.conf. |
| 14:40.03 | kaldemar | and the second is type=user. |
| 14:41.01 | BorjaGVO | How can it be that Asterisk matches ip-port or the "user name in from" from a request before actually knowing what type it is? |
| 14:41.20 | blitzrage | you define the type |
| 14:41.30 | BorjaGVO | yes |
| 14:41.31 | blitzrage | and asterisk seeks when an INVITE comes in for a match |
| 14:41.33 | BorjaGVO | I know that |
| 14:41.41 | BorjaGVO | yeah |
| 14:41.42 | BorjaGVO | right |
| 14:41.49 | BorjaGVO | but... |
| 14:42.12 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 14:42.32 | BorjaGVO | how does Asterisk know what match against (ip-port or name in from)... |
| 14:42.40 | BorjaGVO | ? |
| 14:42.53 | [TK]D-Fender | ... |
| 14:42.57 | blitzrage | it has an order that it tries to match again |
| 14:43.00 | blitzrage | st |
| 14:43.08 | bchia | When Asterisk gets a SIP packet the first thing it looks at is username in the from: line of the SIP header.. |
| 14:43.09 | blitzrage | it tries to match username first, then IP address |
| 14:43.15 | [TK]D-Fender | If you HAVE a user, it tries your user. If you don't, and have a peer, it looks for your peer |
| 14:43.15 | blitzrage | bchia: that |
| 14:43.17 | blitzrage | :) |
| 14:43.22 | bchia | it then tries to match that against a user/Friend in the sip.conf cache |
| 14:43.24 | kaldemar | BorjaGVO: http://svn.digium.com/svn/asterisk/tags/11.1.2/configs/sip.conf.sample <-- "Naming devices" |
| 14:43.26 | blitzrage | so it says, "ok, do I have any type=user I can match again" |
| 14:43.29 | blitzrage | if not, then it moves onto peer |
| 14:43.36 | bchia | if that's not found then it lookd at IP and tries to match on type=peer |
| 14:44.06 | BorjaGVO | Let me read you guys ;-) |
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| 14:48.34 | BorjaGVO | Ok, so there is no differences between them when making/receiving calls? I mean, if THAT is only the difference (priority of looking into sip.conf), I don't understand the actual "difference" |
| 14:48.52 | BorjaGVO | Do you know what I mean? |
| 14:49.04 | bchia | consider a phone that has mutiple accounts registered to it - |
| 14:49.14 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 14:49.15 | bchia | you CAN'T use "peer" because it matches on IP |
| 14:49.22 | bchia | you need to use friend |
| 14:49.42 | blitzrage | (which is a short form of a peer and user) |
| 14:50.16 | bchia | blitzrage is right - literally in the guts a friend is a peer and user together (but in regards to matching it behaves like a user) |
| 14:50.19 | kaldemar | and a peer really is useful for practically any scenario. |
| 14:50.45 | blitzrage | bchia: which is because the user is matched first prior to attempting matching on the peer structure |
| 14:50.47 | blitzrage | <--- leifmadsen btw |
| 14:50.52 | blitzrage | bchia: if you were unaware |
| 14:50.58 | blitzrage | kaldemar: +1 |
| 14:51.03 | bchia | hey lief :) (didn't know) |
| 14:53.43 | bchia | is "blitzrage" a reference to Final Fantasy X at all? |
| 14:54.45 | jacekowski | strange thing on ISDN2e (BRI) lines from BT with 1.8.11-cert8 and 2.6.1 dahdi |
| 14:55.01 | jacekowski | after restart i can't make any outgoing calls |
| 14:55.14 | *** join/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 14:55.15 | jacekowski | and then as soon as call comes in from the outside, everything works fine |
| 14:55.34 | bulkorok | sounds like the sync is not done... |
| 14:55.38 | BorjaGVO | Thanks guys, much clearer! :) |
| 14:55.59 | gavimobile | folks, I want to move my cdr mysql database to a remote location. what files should be changed other than nano /etc/odbc.ini? |
| 14:56.11 | BorjaGVO | kaldemar: why is it peer useful for practically any scenario? |
| 14:56.40 | kaldemar | BorjaGVO: because it matches both by the username and IP/port. |
| 14:56.50 | *** join/#asterisk Naikrovek (c02b41f5@gateway/web/freenode/ip.192.43.65.245) |
| 14:56.52 | bulkorok | jacekowski: setup a sipaccount and ring your ISDN after restart with asterisk |
| 14:57.07 | BorjaGVO | kaldemar: you mean type=driend? |
| 14:57.26 | kaldemar | BorjaGVO: no, i mean type=peer, just like i said. :) |
| 14:57.57 | *** part/#asterisk Naikrovek (c02b41f5@gateway/web/freenode/ip.192.43.65.245) |
| 14:59.34 | BorjaGVO | kaldemar: then peer is the same as friend? hehe...I might didn't catch it as well as I thought |
| 14:59.56 | teloniusz | jacekowski: it sounds like kewlstart/loopstart problem, but that would be typical for analog lines... |
| 15:01.32 | teloniusz | jacekowski: what board do you use? (output from lspci and dahdi_scan would be useful) |
| 15:03.38 | jacekowski | 05:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) |
| 15:03.50 | jacekowski | openvox clone of digium hardware |
| 15:04.07 | kaldemar | BorjaGVO: not, not at all same as friend, as "friend" makes a type=user and type=peer. |
| 15:04.35 | Katty | infinity1: crittercam |
| 15:04.36 | Katty | oh |
| 15:04.40 | Katty | infinity1: disregard! |
| 15:04.43 | Katty | infobot: crittercam |
| 15:04.44 | infobot | i heard crittercam is Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
| 15:04.46 | Katty | ^- squirrel! |
| 15:06.17 | BorjaGVO | kaldemar: ok, and why one would want to be peer and user at the same time? Why not being just user as it is more specific? |
| 15:07.22 | kaldemar | BorjaGVO: they haven't always worked in the manner they do nowadays. |
| 15:09.15 | BorjaGVO | ok, so the existance of all three types is maintained because prior versions? Could everything work fine just with friend right? I don't find a case where it's better to use peer. Any? |
| 15:09.28 | Katty | HAPPY FRIDAY! |
| 15:10.33 | [TK]D-Fender | <BorjaGVO> kaldemar: ok, and why one would want to be peer and user at the same time? Why not being just user as it is more specific? <- because it depends how the other side sends you calls vs how you have to send THEM calls. |
| 15:10.55 | [TK]D-Fender | BorjaGVO, If their IP may vary then PEER will not work. |
| 15:12.04 | [TK]D-Fender | BorjaGVO, If you have multiple phones at a give IP and need to ID a call FROM there, then those phone's entries should ALSO not be PEER. Because with the IP all matching it'll just hit the FIRST entry and fail auth. |
| 15:12.17 | kaldemar | unless they re-register when their IP changes. |
| 15:12.27 | [TK]D-Fender | BorjaGVO, Phones behind the same NAT would therefor normally be type=friend which acts as both |
| 15:12.50 | [TK]D-Fender | kaldemar, those phones all share the same WAN however |
| 15:12.54 | [TK]D-Fender | kaldemar, So not applicable |
| 15:15.40 | bchia | type=user can't make outbound calls (it's inbound only) - when you pass an argument to the dial it has to be a type=peer or type=friend in sip.conf e.g. Dial(SIP/peer-name) |
| 15:15.50 | jacekowski | another weird thing, it has never happened before, asterisk kept running, but would not accept any calls or anything |
| 15:15.55 | BorjaGVO | [TK]D-Fender, kaldemar, @blitzrage, bchia: Thanks very much. I appreciate your help. I think now I got it clear. :) |
| 15:17.33 | BorjaGVO | bchia, what is the reason because type=peer can not make outbound? I guess it is just designed like that, right? |
| 15:18.00 | bchia | type = peer CAN make outbound / type = user cannot |
| 15:18.39 | BorjaGVO | yes, yes, sorry..that is what I meant |
| 15:19.06 | bchia | I'm not sure of the reasoning behind the configuration (I suspect we have what we have today because it evolved that way) but this is the behavior that we have today |
| 15:19.56 | BorjaGVO | Alright, so summarized: |
| 15:20.31 | bchia | I think that is a long-forgotten time user was for inbound only, peer was for outbound only and friend was for doing both inbound and outbound (you may still see some of this outdate documentation on the web as this is no longer the case - peer can do both inbound/outbound) |
| 15:20.53 | BorjaGVO | user: CAN'T make calls, it only receives. |
| 15:21.14 | file | that depends upon the direction you view things... |
| 15:21.34 | BorjaGVO | peer: outbound/inbound |
| 15:21.57 | BorjaGVO | friend: outbound/inbound |
| 15:22.08 | BorjaGVO | BUT |
| 15:22.44 | BorjaGVO | peer is not ment for multiple accounts behind one single IP, that is the main difference between peer and friend |
| 15:22.54 | BorjaGVO | is it right? |
| 15:23.05 | bchia | I think that's a good summary |
| 15:23.46 | blitzrage | wait, peer for multiple accounts behind single IP? that sounds backwards |
| 15:23.56 | blitzrage | you mean multiple users for endpoints behind a single IP |
| 15:24.06 | [TK]D-Fender | BorjaGVO, friend is like creating peer & user at the same time |
| 15:24.11 | blitzrage | (wait... not enough coffee -- ports will match for independent) :) |
| 15:24.16 | blitzrage | continue on! |
| 15:24.31 | [TK]D-Fender | BorjaGVO, You COULD do these separately if you wanted to. This just saves you space & time |
| 15:27.50 | BorjaGVO | alright. I'll write this down because I'm sure the doubt will arise sooner or later (in my case for sure :) ) |
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| 15:45.17 | BorjaGVO | Sorry for being so anoying with this, hehe, but type=peer only tries to match IP adress, not "from", right? |
| 15:45.37 | *** join/#asterisk v0lZy (~chatzilla@84-255-194-41.static.t-2.net) |
| 15:48.24 | [TK]D-Fender | correct |
| 15:50.33 | BorjaGVO | [TK]D-Fender, thanks |
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| 15:53.47 | dorphalsig | Hello |
| 15:54.16 | dorphalsig | I just upgraded my asterisk from 1.8 to 11, however I seem to have broken the t.38 support (spandsp) |
| 15:54.40 | dorphalsig | I'm getting this error: |
| 15:54.53 | dorphalsig | WARNING[2432][C-00000000]: res_fax.c:1662 receivefax_t38_init: channel 'SIP/10.8.45.185-00000000' timed-out during the T.38 negotiation. |
| 15:56.12 | dorphalsig | http://pastebin.ca/2299392 |
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| 16:01.49 | bombev | have a good one |
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| 16:11.59 | dorphalsig | Hello? |
| 16:13.47 | *** join/#asterisk vjfromgt (~vjfromgt@pool-96-224-208-204.nycmny.fios.verizon.net) |
| 16:14.45 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 16:14.50 | vjfromgt | is it possible to overide caller id on a trunk level (outbound) ie, regardless of what the calelr id is , the trunk config always send a static caller id |
| 16:25.03 | kaldemar | vjfromgt: in dialplan with app Set and func CALLERID before the Dial. |
| 16:26.11 | kaldemar | vjfromgt: fromuser in sip.conf might also be what you're looking for if you use SIP. |
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| 16:48.12 | dorphalsig | I just upgraded my asterisk from 1.8 to 11, however I seem to have broken the t.38 support (spandsp). I've searched on voip-info wiki and some forums, but have gotten no answer. I already installed spandsp, however I keep getting these messages. http://pastebin.ca/2299392 |
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| 16:58.09 | *** join/#asterisk pcAngel (yoink@S0106c8be195a474c.vc.shawcable.net) |
| 17:03.48 | dorphalsig | I just upgraded my asterisk from 1.8 to 11, however I seem to have broken the t.38 support (spandsp). I've searched on voip-info wiki and some forums, but have gotten no answer. I already installed spandsp, however I keep getting these messages. http://pastebin.ca/2299392 |
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| 17:25.09 | moos3 | anyone send 1.6.2 eat up memory like mad ? My asterisk install is using 7G's of memory |
| 17:26.25 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
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| 17:35.55 | pcAngel | hopefully someone here knows the asterisk source code... |
| 17:36.31 | [TK]D-Fender | pcAngel, Some people know certain parts of it. Ask your specific question and we'll see if we have a specific answer for you. |
| 17:36.40 | pcAngel | I'm in process_sdp right now, I'm trying to figure out if the variable "p" or the variable "req" represents the side sending us a sdp session invite |
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| 17:36.43 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:37.10 | file | p is the SIP dialog, req is the SIP request received |
| 17:37.29 | pcAngel | Is there a way to get the IP address it was received from, from req? |
| 17:37.57 | pcAngel | I have a client who's router is sending the IP address "47.145.127.9206" (extra 06 at the end) causing getaddrinfo in process_sdp_c to fail |
| 17:38.32 | file | it may be in req, I don't remember |
| 17:38.37 | pcAngel | The only solution I can think of is to have it automatically fall back to the IP address it was sent from |
| 17:38.45 | pcAngel | ok I'll look it up in the headers |
| 17:39.04 | pcAngel | it's strange that nat=yes isn't coming in to play anywhere in here |
| 17:39.25 | file | it doesn't come into play until traffic is actually received |
| 17:39.27 | pcAngel | do you know if there's a reason for that? it makes me feel like I'm playing with fire to bypass it |
| 17:39.35 | pcAngel | ok |
| 17:39.41 | file | it still uses the provided IP address/port until that time |
| 17:40.14 | *** join/#asterisk camelCase (~camelCase@unaffiliated/camelcase) |
| 17:40.16 | camelCase | hello |
| 17:40.43 | camelCase | I am having some trouble getting media working with webRTC |
| 17:41.05 | camelCase | has anyone successfully set up * with webRTC support? |
| 17:41.29 | file | I have. |
| 17:41.32 | camelCase | When I place a call the audio cuts out when I pick up |
| 17:41.47 | camelCase | file, would you mind helping me out a bit? |
| 17:42.05 | file | there's nothing I can explicitly help with, Chrome is very much still in flux |
| 17:42.20 | file | it was broken, then worked, then they broke it again, then I think they fixed it |
| 17:42.21 | camelCase | i realize that |
| 17:42.45 | camelCase | :/ |
| 17:42.57 | camelCase | so it is not necessarily * ? |
| 17:43.14 | file | I haven't modified the WebRTC code since I put it in *months* ago |
| 17:43.33 | file | almost all of the problems have been with the WebRTC support in Chrome |
| 17:43.52 | camelCase | I recompiled 11 with the right options and configured my sip.conf to the specs listed on the * site |
| 17:43.58 | camelCase | ugh |
| 17:44.10 | file | it's still being developed and the specification still being done |
| 17:44.24 | camelCase | I am a VoIP security consultant |
| 17:44.33 | camelCase | I like to stay on the bleeding edge |
| 17:44.40 | camelCase | sometimes is it uncomfortable |
| 17:44.42 | camelCase | :) |
| 17:45.18 | camelCase | but you had it working at some point with the * configs per the wiki? |
| 17:45.24 | file | yes. |
| 17:45.45 | camelCase | see, i was thinking it was a codec issue |
| 17:46.10 | camelCase | did you run that patch for STUN? |
| 17:46.15 | file | nope |
| 17:46.21 | file | I have always used unpatched Asterisk |
| 17:46.41 | camelCase | it didn't work before or after |
| 17:46.58 | file | these last few times it has been an ICE or SDP problem |
| 17:47.11 | camelCase | SDP seems likely |
| 17:47.27 | camelCase | although I am not sure how the webrtc signaling works |
| 17:47.51 | camelCase | but if it is a browser issue maybe it will work soon |
| 17:48.13 | camelCase | i hate going into chat rooms and asking for help this was kind of a last ditch hope |
| 17:48.20 | camelCase | thanks for the input |
| 18:01.24 | ghost75 | if i want to park a call over ami, what to specify as target channel? |
| 18:07.11 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 18:11.38 | pcAngel | Thanks for the help file, I'm going to have to finish it later but I guess I need to pass the origin ast_sockaddr from handle_request_invite to process_sdp and then process_sdp_c... PITA |
| 18:12.27 | pcAngel | to memcpy it and return the origin sockaddr when the IP in the sip dialog is invalid |
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| 18:13.04 | *** part/#asterisk Primer (~daniel@www.ceregatti.org) |
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| 18:35.02 | *** join/#asterisk citrusfizz (~chatzilla@70.184.40.66) |
| 18:36.49 | citrusfizz | can anyone recommend a "HD Voice" hard phone? would be used in our conference rooms, and i would need to configure our asteriskNOW 2.0.2 installation for it most likely |
| 18:42.26 | [TK]D-Fender | citrusfizz, Who is going to be on the other end? |
| 18:45.55 | pabelanger | citrusfizz, what IP phones do you use now? |
| 18:46.09 | citrusfizz | currently using cisco IP phone 7960 |
| 18:46.12 | citrusfizz | in sip mode |
| 18:46.53 | citrusfizz | these phones are starting to show their age, and i have had to replace a few recently, so instead i would like to look towards an upgrade |
| 18:47.17 | pabelanger | Polycom |
| 18:47.22 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:48.07 | Qwell | citrusfizz: Digium phones work very well with AsteriskNOW. |
| 18:48.21 | Qwell | oh, conference phone. meh |
| 18:49.13 | jpsharp | Not to be a Debby Downer, but unless you're using an HD codec from end to end, an "HD Voice" phone isn't going to do you a bit of good. |
| 18:49.26 | Qwell | sure it will |
| 18:50.03 | Qwell | They have higher quality speakers/mic, so even G.711 sounds better. |
| 18:50.16 | citrusfizz | well, if we replace all the office phones with a good model, then interoffice communication will be improved at the least |
| 18:50.57 | Qwell | citrusfizz: I definitely recommend Digium phones for that. I'm not biased at all, having written AsteriskNOW and the FreePBX interface for our phones that it ships with. |
| 18:51.18 | citrusfizz | Qwell: are the digium phones good for speaker calls (conference) |
| 18:51.44 | Qwell | If you're going to have it in a conference room, you definitely want a "real" conference phone. The speaker phones are very good though. |
| 18:52.28 | citrusfizz | whats a REAL conference phone |
| 18:52.46 | Qwell | like a Polycom SoundStation |
| 18:53.35 | Qwell | multiple mics, etc |
| 19:06.51 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:11.47 | citrusfizz | hmm the soundstation ip 5000 might seem like a good price point |
| 19:14.43 | *** join/#asterisk acedia (~rage@unaffiliated/ffs) |
| 19:16.40 | rrittgarn | +1 to Qwell |
| 19:17.54 | rrittgarn | we sell the 5000's as well as the 7000s, if you don't need the remote mics, the 5000s are nice phones for small conference rooms |
| 19:18.13 | _Corey_ | I think they're doing away with the 5000s for whatever it's worth |
| 19:30.23 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 19:32.29 | Docfxit | I would like to record a voice prompt. Under what heading in the extensions.conf should I put the new extension? |
| 19:36.07 | Qwell | Docfxit: where ever it makes sense, given your specific scenario. |
| 19:37.53 | carrar | Under the heading ;; Voice Prompt that I recorded |
| 19:38.14 | Docfxit | Thanks. |
| 19:40.18 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:40.22 | ghost75 | (19:02:03) ghost75: if i want to park a call over ami, what to specify as target channel? |
| 19:40.54 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:44.45 | jpsharp | The channel where the call you want to park resides. |
| 19:46.44 | jpsharp | First channel is the channel to park, second channel is the channel to play parking slot information to. |
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| 19:53.26 | jacekowski | citrusfizz: i'm using digium phones |
| 19:55.05 | jacekowski | paging question, i want overhead paging rather than paging through the speakers in the phones |
| 19:55.17 | jacekowski | and i want to reuse as much as possible from the old system |
| 19:55.50 | jacekowski | which is basically, active speakers with analogue signal + power coming in |
| 19:56.24 | jacekowski | what would be my best option when it comes to spending least amount of money and time setting it up |
| 19:56.43 | jacekowski | and why does SIP->PA gateways cost 10x as much as a normal sip phone |
| 19:56.56 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:58.53 | _Corey_ | jacekowski: Probably because the market is pretty limited... you can just buy an FXO/FXS port and plug into just about any analog paging system without changing much |
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| 20:08.45 | Docfxit | I tried to restart asterisk with sudo asterisk -r I received an error saying Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) The file does exist currently with zero bytes. How can I restart asterisk ver. 1.4.22 without rebooting the machine? |
| 20:08.48 | blitzrage | jacekowski: ATA + Bogen UTI1 |
| 20:08.55 | blitzrage | jacekowski: most amazing setup ever for overhead paging |
| 20:09.45 | blitzrage | jacekowski: the ATA plugs into your network (SIP side), the analog side plugs into the Bogen, and the bogen plugs in and controls the amplifier |
| 20:10.08 | blitzrage | the Bogen is basically the switch to determine when to allow audio from the ATA into the paging system via a dialtone/ringing signal |
| 20:10.19 | blitzrage | works amazing -- I've done it in several locations |
| 20:10.53 | blitzrage | jacekowski: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/AdditionalConfig_id257169.html |
| 20:11.07 | blitzrage | read External Paging section |
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| 20:13.36 | reisi | in dialplan, should Page application block even after everyone paged has answered? |
| 20:13.50 | blitzrage | it's like a Dial() |
| 20:15.19 | file | DialAndBroadcast() |
| 20:18.57 | reisi | so, is there a way to make it non-blocking, as in continue to execute following dialplan with the meetme setup? |
| 20:20.57 | [TK]D-Fender | reisi, All dialplan is "blocking" |
| 20:22.09 | reisi | hmmm strange, i'm pretty sure i have had it previously working as such, perhaps it was a bug then? |
| 20:24.05 | [TK]D-Fender | No. |
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| 20:27.49 | reisi | so no way to get past the Page application until the conference is destroyed (caller hangups) |
| 20:29.15 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 20:30.42 | [TK]D-Fender | reisi, When would you expect or want that to happen, and to what end? |
| 20:31.01 | reisi | i was hoping to Playback(...) a file for everyone joined (after the timeout passes or everyone has joined), any ideas on how this could be implemented, if Page blocks? |
| 20:31.46 | [TK]D-Fender | reisi, You should rethink how you're doing this. |
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| 20:49.20 | reisi | [TK]D-Fender: hmmm ok, but still cannot see the solution; i'm trying to playback a file simultaniously to all parties, what could be the right way to solve this? |
| 20:49.25 | file | Page has an 'A' option which takes a sound file to play to all paged devices |
| 20:49.55 | reisi | file: just like Playback? |
| 20:50.38 | file | in the sense that it plays back a sound file, sure |
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| 21:10.10 | [TK]D-Fender | reisi, First you need to think about how you're triggering this in the first place |
| 21:10.48 | [TK]D-Fender | reisi, Dialplan is linear. You want the calling end to do a playback then you're looking at having a LOCAL channel on one side and app_page on the other |
| 21:11.22 | [TK]D-Fender | reisi, Then again Page has a variable set up time, and you're better off spawning SEPARATE local channels to playback to each independently |
| 21:11.32 | [TK]D-Fender | reisi, As dialplan length is also limited |
| 21:11.37 | [TK]D-Fender | for the actual call. |
| 21:14.53 | rrittgarn | reisi: are all parties picking up simultaneously? You could use local channels to call Dial(Local/Number1@PlaySoundContext&Local/Number2@PlaySoundContext) and in the command in your PlaySoundContext use the 'g' option to continue with dialplan even if the other party hangs up, so all your users get the message. (If i understand what you're going for) |
| 21:23.39 | reisi | rrittgarn: yes, they pick up simultaneously |
| 21:24.19 | reisi | thanks for pointers again, both of you, learning local channels next |
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| 22:32.57 | hugo | hey, I'm setting up * for testing and learning and have a group of friends using it with me. what would I set to only allow calls between us, ie only on local domain(s) ? |
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| 22:33.24 | jmetro | only setup outbounds allowed to the number of #'s in your extensions? |
| 22:33.50 | jmetro | unless they are modifying dialplan too |
| 22:34.39 | hugo | sadly those terms don't mean anything me yet |
| 22:34.40 | hugo | lol |
| 22:34.54 | hugo | sorry, I'm really new to the whole thing still |
| 22:35.15 | WIMPy | ~book |
| 22:35.15 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:35.15 | jmetro | basically you can only dial out to what you code our dialplan to do, so if you only code dialplan to other numbers inside your network, that is all you will ever hit |
| 22:36.09 | hugo | hmm |
| 22:37.05 | hugo | also, is it possible to run tls over udp port 5061? |
| 22:37.56 | WIMPy | Asterisk doesn't do TLS over UDP. |
| 22:38.14 | hugo | ok |
| 22:40.59 | jmetro | following the book is the best way to learn asterisk. just read it all and then follow along when it starts installing linux and then by the end of a couple chapters you have working dialplan and enpoints |
| 22:41.36 | hugo | I will |
| 22:41.52 | hugo | for the past few days I was setting it up just by reading the config but then ran into a trouble with the netbsd libssl and the rtp module |
| 22:42.04 | hugo | and finally sorted that out today (well, some guy in the mailing list did) |
| 22:43.23 | hugo | anyway guys I have to dash now, will be online again soon |
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| 23:13.23 | *** join/#asterisk raden (~Jon@24-240-51-238.dhcp.stpt.wi.charter.com) |
| 23:13.33 | raden | anyone run SMS on asterisk |
| 23:13.48 | raden | hugs katty !!!!!!!!!!!!!!!!!!!!!!!! |
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| 23:15.47 | ghost75 | (20:47:24) jpsharp: First channel is the channel to park, second channel is the channel to play parking slot information to. <- but only first channel exists |
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| 23:57.34 | *** join/#asterisk schultza (~schultza@rc1.rcherbals.com) |
| 23:58.31 | schultza | im following the asterisk - the definitive guide and it's recommending to install libpri before dahdi... but im getting an error while following instructions. does dahdi need to be downloaded for libpri to install... or does dahdi need to be installed before libpri is installed? |
| 23:58.38 | schultza | svn co downloaded |