| 00:19.29 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:30.11 | *** join/#asterisk mattjackets (~matt@c-71-236-96-210.hsd1.pa.comcast.net) |
| 00:31.46 | mattjackets | asterisk 1.6 not trying to register with sip provider. i've used asterisk for years, just upgraded to 1.6 on debian (from 1.4) and asterisk is not trying to register with gizmo5 via sip. seems like register => line is ignored. help pls :) |
| 00:32.43 | mattjackets | verbosity is huge (68) and sip debug is on |
| 00:32.53 | mattjackets | no registration attempts are shown |
| 00:34.22 | p3nguin | mattjackets: What version of Asterisk are you using? |
| 00:34.36 | mattjackets | 1.6.2.9 |
| 00:35.07 | p3nguin | I would first recommend upgrading to the current version in your chosen branch. |
| 00:35.47 | mattjackets | i've been tempted to try 1.8, but I like to stick with the standard debian packages to ease administration (not the case this time) |
| 00:36.10 | p3nguin | Also, since when does Gizmo5 allow SIP registration? |
| 00:36.18 | mattjackets | do you know of any issues regarding sip registration in 1.6? has there been a major change since 1.4? |
| 00:36.25 | mattjackets | it always has |
| 00:36.27 | TeknoJuce | can't hurt to try and then go back when it doesnt work |
| 00:37.01 | p3nguin | I've never known Gizmo to allow registrations. That's news to me. |
| 00:37.24 | mattjackets | yeah, that's how i've used it for years with google voice... |
| 00:37.34 | SunTsu | mattjackets: I upgraded asterisk 1.2 to asterisk-1.6.2.16.1 recently, sip registry worked immediately, only the dialplan really needed some work |
| 00:38.24 | mattjackets | thanks suntsu....it didn't look like things changed, but it's good to know |
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| 00:39.43 | p3nguin | I doubt 1.6.2.9 has broken registration, so show me your register statement, masking only the password. |
| 00:40.37 | mattjackets | register => 1747635xxxx:password@proxy01.sipphone.com |
| 00:41.35 | p3nguin | Can you register to other services? |
| 00:42.11 | mattjackets | later i have a 'friend' defined as [gizmo] with context, insecure=port,invite, secret, defaultuser, fromuser,fromdomain and a few other options set |
| 00:42.27 | mattjackets | no, i had a couple others defined and they don't work either |
| 00:43.14 | p3nguin | That's really strange. I assume you can't run "sip set debug on" if the sip module isn't working. |
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| 00:44.43 | mattjackets | i can set debug on |
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| 00:45.20 | mattjackets | i can see the sip stuff in sip show peers, and my hard phones (sip) can register to asterisk |
| 00:45.45 | p3nguin | What does "sip show registry" show? |
| 00:46.07 | mattjackets | nothing at all :( |
| 00:46.25 | p3nguin | Literally nothing, or just no peer/registry entries? |
| 00:47.01 | mattjackets | sorry...here's what it shows: |
| 00:47.10 | mattjackets | Host dnsmgr Username Refresh State Reg.Time |
| 00:47.10 | mattjackets | 0 SIP registrations. |
| 00:47.44 | p3nguin | The current version works, so you should consider upgrading to it. Or at least try a different version. |
| 00:48.02 | p3nguin | Current in 1.6.2 branch is 1.6.2.17.2. |
| 00:48.09 | mattjackets | ok, thanks for the help p3nguin |
| 00:48.45 | p3nguin | If you're worried about not having a package to install if you build from source, just use checkinstall to install from source. |
| 00:49.48 | p3nguin | Instead of "make install" as your last command to install, use "checkinstall -D" to build a debian package and install it. Now you'll keep package manager consistency. |
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| 00:51.11 | mattjackets | p3nguin: i've done that before, but it's just so much easier to let someone else package security fixes for me :) this is a home server, and honestly i can't give it much attention |
| 00:52.16 | p3nguin | I just don't know why registration wouldn't work on that version. I'm sure it worked for other people using that same version. |
| 00:52.35 | mattjackets | oh, yeah...no kidding! |
| 00:52.47 | p3nguin | You had the register statement in place when you started asterisk, right? |
| 00:53.34 | mattjackets | yep |
| 00:53.52 | p3nguin | Very odd. |
| 00:54.46 | mattjackets | i agree....i think i'll wipe out my config and start from scratch....one step at a time |
| 00:54.57 | mattjackets | i hope it's a config issue and not the package |
| 00:55.31 | p3nguin | Your register statement is in the [general] section of sip.conf, isn't it? |
| 00:55.55 | p3nguin | It needs to be under [general] and before [authenticate]. |
| 00:56.18 | p3nguin | Err... [authentication]. |
| 00:58.26 | mattjackets | p3nguin! you are the best! |
| 00:58.38 | p3nguin | You had it placed incorrectly? |
| 00:58.50 | mattjackets | i didn't notice the authentication section in the 1300 line config file |
| 00:59.15 | p3nguin | 1300 lines? You certainly don't need that much stuff in the sip.conf. |
| 00:59.23 | mattjackets | i did! i wanted to keep all my changes in one place since the file was soooo big. i just assumed [general] was the only thing in there |
| 00:59.54 | p3nguin | You should never use the sample config as an actual config. |
| 00:59.55 | mattjackets | it's mostly comments......i can't stand it. it's like the package maintainer just dumped the manpages in there |
| 01:00.33 | mattjackets | i didn't for extnesions, but just kinda went with it for sip.conf. lesson learned :) |
| 01:00.49 | p3nguin | Eh, extensions go in a different file (extensions.conf). |
| 01:01.51 | mattjackets | yep, extensions.conf is what i replaced....the packaged config had way too much stuff in it for me to be comfortable |
| 01:02.38 | mattjackets | thanks again p3nguin! i really appreciate it |
| 01:02.44 | p3nguin | So the packages have .conf files rather than .conf.sample or .conf.default or something to indicate that they are SAMPLE files? |
| 01:03.31 | p3nguin | If so... What a horrible way to package Asterisk. |
| 01:03.43 | mattjackets | yes. they put them in place, and even start the server. |
| 01:03.54 | p3nguin | That's ridiculous. |
| 01:03.59 | mattjackets | it's primarily a demo, but it connects to digium & stuff |
| 01:04.05 | p3nguin | I see. |
| 01:04.22 | p3nguin | Maybe that sort of makes sense. |
| 01:04.25 | p3nguin | Maybe. |
| 01:04.30 | mattjackets | yeah, especially when they update the config files in the package and apt wants to mess with your custom config |
| 01:04.32 | p3nguin | I'm not sure. |
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| 01:05.03 | p3nguin | I'll be sure to add this to my list of reason why I will never use Debian nor anything that looks like Debian. |
| 01:05.07 | mattjackets | other debian packages put sample files in /usr/share/doc....and you have to copy them out by hand if you want to base your setup on them |
| 01:05.21 | mattjackets | aw :( i don't like you anymore |
| 01:05.24 | p3nguin | That's sensible. |
| 01:05.38 | mattjackets | what's your distro of choice? |
| 01:06.51 | p3nguin | I primarily use Arch Linux, but also use FreeBSD, OpenBSD, CentOS, and various other obscure OSs. |
| 01:08.40 | mattjackets | is arch the distro that gives you a minimalistic compilation environment and nothing else? no package manager, just build from tars? |
| 01:08.51 | p3nguin | No. |
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| 01:09.07 | p3nguin | You're probably thinking of either LFS or Gentoo. |
| 01:09.33 | mattjackets | kinda like lfs....but a little more structure to get you started. |
| 01:09.45 | p3nguin | I'm not sure what that would be. |
| 01:09.48 | mattjackets | not as much structure as gentoo.....maybe it's called core linux |
| 01:10.07 | p3nguin | Arch has a pretty good package manager. |
| 01:10.53 | p3nguin | It also provides a wonderful repository for user contributed software. |
| 01:10.58 | mattjackets | i'm a debian fan....for the most part i think they do things very well....but sometimes things like this come out of the woodwork |
| 01:13.06 | p3nguin | Arch core install doesn't have a lot to it, and a lot of newbs can't figure out what to do with it, but it provides everything you need to build your OS the way you want it. |
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| 01:14.48 | Shaaan | Is there a way when i dial a line im prompted to enter callerID 10 digits then the phone number so then when i call that number it displays that callerID that i input? |
| 01:15.09 | p3nguin | Sure. |
| 01:15.25 | Shaaan | how could i do that example please? |
| 01:15.33 | p3nguin | One moment. |
| 01:21.57 | p3nguin | shaaan: Still here? |
| 01:22.45 | Shaaan | ye |
| 01:22.48 | Shaaan | *yes |
| 01:23.03 | p3nguin | shaaan: http://pastebin.com/baETyjJG |
| 01:23.32 | p3nguin | Disclaimer: This paste is not checked for typos/bugs. |
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| 01:26.25 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 01:27.09 | saliak | p3nguin: sorry, i left before you responded. what do you mean by ipcomms? |
| 01:27.31 | p3nguin | saliak: Hmm? |
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| 01:32.00 | p3nguin | shaaan: Are you having any trouble implementing that? |
| 01:32.08 | Shaaan | looking at it |
| 01:32.45 | p3nguin | some-context simply represents a context that your phone can dial numbers in. |
| 01:32.58 | p3nguin | [phones] for example |
| 01:33.31 | p3nguin | outbound-calls represents your existing outbound calling context. |
| 01:34.07 | saliak | p3nguin: sorry, i left before you responded. what do you mean by "Rhode Island? Must be IPcomms"? |
| 01:34.32 | p3nguin | You can modify your already-existing outbound calling context by adding the (nocid) label on your outbound Dial(). |
| 01:35.52 | juliocesarlhg | i would like to know how to .net with asterisk |
| 01:35.58 | p3nguin | saliak: I don't know who was involved in the conversation or anything, but I saw a phone number mentioned which was in the 401 area code, and I figured it was a Rhode Island number. Then I had an idea that the provider was IPcomms. |
| 01:35.58 | juliocesarlhg | c# |
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| 01:36.14 | p3nguin | juliocesarlhg: #asterisk-dev |
| 01:39.27 | saliak | p3nguin: Yeah, i'm trying to get my incoming sip trunk working. I am in RI, but i'm using broadvoice (i'm not sure if they're a subset of ipcomms or what). For some reason the outgoing is working OK, but my incoming doesn't seem to answer. When i turn on sip debug it spits out alot of stuff when the call comes in, but there seems to be something with the extension (well, that's my diagnosis, probably wrong...) debug - |
| 01:39.28 | saliak | http://pastebin.com/06EESiFu, sip - http://pastebin.com/cJzTGWWb, extensions - http://pastebin.com/d4yBGzzX |
| 01:40.18 | p3nguin | pastes seem to have expired. |
| 01:40.23 | p3nguin | wait |
| 01:40.33 | p3nguin | false alarm |
| 01:41.50 | saliak | it keeps telling me "the party you are trying to reach is unavailable". i feel like i must be missing something basic |
| 01:41.56 | p3nguin | Okay, a few things I'd like to mention... |
| 01:42.25 | p3nguin | You don't need to Answer() before Playback(). |
| 01:42.39 | p3nguin | You should consider not using numbered priorities. |
| 01:43.32 | p3nguin | insecure=very should be insecure=port,invite |
| 01:43.41 | p3nguin | insecure=port,invite should only be used if necessary. |
| 01:44.10 | p3nguin | dtmfmode should really not be inband unless you have a good reason; it should be rfc2833. |
| 01:44.44 | p3nguin | register=4015621302:XXXXX@sip.broadvoice.com/1000 is a failure... |
| 01:45.10 | p3nguin | It should be register => 4015621302:XXXXX@sip.broadvoice.com/4015621302 or register => 4015621302:XXXXX@sip.broadvoice.com |
| 01:45.26 | p3nguin | saliak: I think that's all. Fix those things, then get back to me. |
| 01:45.31 | saliak | cool |
| 01:45.49 | Andrew__M | elb: Are you here? |
| 01:46.47 | p3nguin | If you use register => 4015621302:XXXXX@sip.broadvoice.com you may get calls going to 's' extension. If that happens, then add the /4015621302 to it. |
| 01:47.31 | Andrew__M | I would like to load over 100 entries into AstDB. What is a better way than manual entry? |
| 01:47.44 | p3nguin | andrew__m: shell script |
| 01:48.08 | p3nguin | asterisk -rx 'database put ... ' |
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| 01:49.48 | saliak | p3nguin : so I should get rid of Answer() wholesale? just the playback and hangup? |
| 01:50.43 | p3nguin | saliak: Playback() has an implied Answer() built-in. Unless you are using Playback(some-file,noanswer), you don't need the Answer(). |
| 01:51.29 | Andrew__M | Like 100 phone numbers from a list...? |
| 01:51.53 | p3nguin | andrew__m: Write a shell script. |
| 01:52.35 | p3nguin | Got the phone numbers in a list? Use a loop to read them in. |
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| 01:54.28 | elb | Andrew__M: yes |
| 01:54.49 | elb | oh, looks like your question has probably been answered :-) |
| 01:55.27 | elb | asterisk -rx for every put is probably going to be slowish, but if you're only doing it once ... |
| 01:55.37 | elb | I dunno if you can separate commands by ; |
| 01:55.47 | elb | you might expect it, if you're familiar with expect |
| 01:56.17 | p3nguin | It's not high performance, but it'll work and should be fully portable. |
| 02:01.39 | saliak | p3nguin : blech. now i'm having trouble registering. had this happen last night too.. fixed itself by morning. |
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| 02:09.42 | p3nguin | saliak: Typically, a mistake in a config file such as the one you have does not fix itself after any amount of time. |
| 02:10.34 | p3nguin | Make sure the register => 4015621302:XXXXX@sip.broadvoice.com is after [general] and before [authentication] in your sip.conf. |
| 02:10.36 | saliak | saliak: yeah, i agree. the issue with answering never gets fixed, but for some reason, for the last 3 days, when i futz with this at aroudn this time of night, it starts failing to register |
| 02:11.14 | p3nguin | saliak: The problem with not being able to answer the calls is BECAUSE you aren't registering. |
| 02:11.40 | p3nguin | You aren't registering at all if you don't correct the register statement and make sure it's in the right place in sip.conf. |
| 02:11.53 | saliak | p3nguin: i can make outbound calls (and it says "Registered") most of the time |
| 02:12.09 | p3nguin | saliak: Outbound calls have jack to do with being registered. |
| 02:12.24 | p3nguin | saliak: Registration is for inbound calls only. |
| 02:13.07 | p3nguin | "sip show registry" shows your registrations. If it says registered, make a call to your DID and show me the sip debug of the call. |
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| 02:13.31 | saliak | p3nguin : ah, didn't realize that. ok, so it does at least say "registered" most of the time. turns out that for whatever reason outbound doesn't work for me when it's not registered. maybe just conincedence |
| 02:14.01 | p3nguin | SIP registration is to tell a remote system how to reach you. That's why it's for inbound calling. |
| 02:14.11 | saliak | p3nguin: http://pastebin.com/06EESiFu is a dump of when i called in when it was registered earlier today |
| 02:19.19 | p3nguin | saliak: Your Asterisk is on a public IP address? |
| 02:19.46 | saliak | <PROTECTED> |
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| 02:20.32 | p3nguin | Do you have UDP port 5060 allowed in? |
| 02:22.10 | p3nguin | Maybe pastebin "iptables -L INPUT -nv"? |
| 02:25.55 | saliak | It should be. I use shorewall for my FW. here's the iptables output - http://pastebin.com/PLdKebp4 |
| 02:28.57 | p3nguin | What about "iptables -L net2fw -nv" too? |
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| 02:30.30 | saliak | http://pastebin.com/ZNkvKmXW |
| 02:31.16 | p3nguin | line 13 shows the relevant ports. |
| 02:31.19 | saliak | ACCEPT net $FW udp iax,sip:5063,68,69,rtpstart:rtpstop |
| 02:31.19 | saliak | is the relevant line from my shorewall rules. rtpstart=10000 and rtpstop=20000 |
| 02:31.43 | saliak | yeah, that looks like the translation of that line |
| 02:31.56 | p3nguin | So that's probably not the problem. |
| 02:33.09 | saliak | did the debug output tell you anything? does the fact that it does that much confirm it's not a fw issue? |
| 02:33.52 | p3nguin | The firewall having the correct configuration tells me it's probably not a firewall issue. The sip debug didn't really tell me anything useful. |
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| 02:36.26 | saliak | :( hrm. ok. maybe i should gut out all the extra demo stuff from my sip and extensions file? |
| 02:36.57 | p3nguin | Did you ever fix the register statement? |
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| 02:38.18 | saliak | yeah |
| 02:38.23 | p3nguin | And after you saved the changes to sip.conf, you ran "sip reload"? |
| 02:38.25 | saliak | i fixed it and unfixed it |
| 02:38.29 | saliak | restarted asterisks |
| 02:38.34 | saliak | asterisk |
| 02:38.46 | p3nguin | What does sip show registry say about that peer? |
| 02:39.11 | p3nguin | Wait, you unfixed it? Fix it and leave it fixed. |
| 02:39.48 | p3nguin | register => username:secret@host/extension |
| 02:40.11 | saliak | yeah, sorry, ultimately settled back on fixed |
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| 02:48.33 | saliak | need to sleep. guess i'll see if the morning fixes htis issue again |
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| 05:14.54 | dimm | When I run the call through an analog line, the tube softphone I can not hear beeps and voice subscribers. Then he hangs up, and I hear sirens, "busy ". In the mixmonitor hear and whistle, and a voice subscriber. Do not tell in what could be the problem? |
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| 05:16.12 | KingDavidNYC | hello, can somebody please help me write a sip.conf extension in a realtime table?, this is my first time and one of the fields is not letting me register the phone |
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| 05:18.05 | costal | Hi all |
| 05:19.52 | costal | I'm having some timestamp problems with the cdrs created in asterisk |
| 05:20.05 | costal | and Im just wondering if asterisk get the timestamp from the system |
| 05:26.41 | p3nguin | kingdavidnyc: You're not clear on what you want. Do you want a sip.conf entry or an extension? |
| 05:30.36 | KingDavidNYC | p3nguin: hello p3nguin |
| 05:30.55 | KingDavidNYC | p3nguin: a sip.conf entry |
| 05:32.09 | KingDavidNYC | p3nguin: when I make the entry in the regular si.conf, the phone registers, but when I enable realtime, it doesnt (different entry) leads me to conclude that there is something in the entry that is wrong |
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| 06:27.01 | kaldemar | KingDavidNYC: is realtime working? |
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| 06:34.46 | costal | hi all |
| 06:34.54 | KingDavidNYC | kaldemar: nice, somebody here! |
| 06:35.59 | KingDavidNYC | kaldemar... I am not sure...it says "unable to connect localhost" until I comment the sipusers line |
| 06:37.23 | costal | is there a timezone parameter for cdr_mysql.conf ??? |
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| 06:38.47 | sawgood | what is the 'difference' between logging to /var/log/messages and /var/log/asterisk/full (as stated in /etc/asterisk/logger.conf)? |
| 06:39.55 | kaldemar | KingDavidNYC: there you have it. your realtime setup is not working at all. |
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| 06:41.53 | KNERD | FULL will contain everything..warnings, errors, debug statements, etc |
| 06:41.57 | KingDavidNYC | I checked everything |
| 06:42.38 | KingDavidNYC | kaldemar: extconfig.conf and res_mysql.conf are correct |
| 06:43.03 | KingDavidNYC | kaldemar: mysql working and all users have the permissions |
| 06:43.35 | KingDavidNYC | kaldemar: and if I only leave sippeers in extconfig.conf, it doesn't complaint |
| 06:44.16 | kaldemar | sawgood: there's no mention of /var/log/messages in logger.conf. if you mean /var/log/asterisk/messages and /var/log/asterisk/full, they are just different files that can be configured to contain different output. |
| 06:44.48 | sawgood | kaldemar: thanks .... yes I made a mistake ... but I'm glad you understood |
| 06:47.12 | sawgood | so maybe you want to log debug stuff in one file and everything else in another? |
| 06:49.54 | KNERD | * does that. it has a log level to select. The higher you go, the different file the logs gets filled with more stuff. |
| 06:50.09 | KNERD | log level 4 is gihest thus goes to FULL |
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| 06:51.06 | KNERD | but yes you can select what you want to go into those files |
| 06:51.07 | KNERD | http://www.voip-info.org/wiki/view/Asterisk+config+logger.conf |
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| 06:51.55 | KNERD | it is best to uncomment the higher levels only when needed else you will start getting a lot of very large files |
| 06:54.16 | *** join/#asterisk drdru (~Adium@76.77.182.145) |
| 06:54.33 | drdru | does asterisk have voice detection capability? |
| 06:54.57 | drdru | and is it possible to record and stream a recording to an external ASR service? |
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| 06:57.04 | KNERD | not "voice" detection but silence |
| 06:57.29 | KNERD | i am sure it is if you want to do some coding |
| 06:58.14 | coppice | voice detection is still a patent minefield |
| 06:59.53 | sawgood | I guess I could do something like log all notice,error,warning locally in /var/log/asterisk/full (and the send all WARNINGS to a central syslog server) |
| 07:00.30 | KNERD | yes, if that is yoru wish, you can do that |
| 07:02.00 | TeknoJuce | voice detection mmmm http://custom3dgraphics.deviantart.com/art/Hip-Hurts-v-2-22995489?fullview=1 |
| 07:04.39 | KNERD | isn;t there a new codec in development for broader frequency in the voice band, beisdes the 0-4Khz? |
| 07:05.06 | TeknoJuce | Think its called real-life |
| 07:05.16 | KNERD | nope |
| 07:05.41 | TeknoJuce | didnt think you could get broader frequency than real life |
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| 07:06.29 | KNERD | "Polycom HD VoIP phones sample between 150 Hz and 7,000 Hz," |
| 07:07.11 | TeknoJuce | I like the frequency of texting better... |
| 07:07.13 | KNERD | g711 300 Hz to 3,400 Hz |
| 07:08.09 | kaldemar | KNERD: G.722 |
| 07:08.21 | TeknoJuce | I still think hdtv is a waste of b/w so I would assume hd phone would be overkill as well |
| 07:08.50 | KNERD | "HD VoIP phones can more accurately produce sound thats in keeping with human speech. That in turn makes it easier for you to understand the speaker on the other end, especially for voices in the extreme low or high end or when someone is speaking with an accent thats difficult to understand." |
| 07:08.54 | KNERD | you were saying? |
| 07:09.22 | KNERD | G.722...hmmm |
| 07:09.43 | TeknoJuce | The only accent we get over here is french and nobody can understand them anyways |
| 07:10.45 | TeknoJuce | wouldnt lie that on the phones fault |
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| 07:11.18 | KNERD | kaldemar: people seem to think G.722 is too outdated |
| 07:13.48 | KingDavidNYC | Somebody please tell me what I am setting up wrong in this f#$%#$% realtime setup |
| 07:15.08 | KingDavidNYC | I am compiling 1.6.2 and still it says "invalid database specified" and I have the correct one!! |
| 07:15.56 | KNERD | why not go for 1.8? |
| 07:16.39 | SiNGLer | KingDavidNYC: did you specify profile name in extconfig.conf? (not the database name) |
| 07:16.59 | kaldemar | KingDavidNYC: no one will be able to tell you anything before you show something. |
| 07:17.24 | KingDavidNYC | kaldermar: what do you need to see? |
| 07:17.42 | KingDavidNYC | Singler: profile name? |
| 07:18.01 | SiNGLer | KingDavidNYC: show your database config and extconfog.conf |
| 07:18.42 | KingDavidNYC | I need to post it in pastebin right? |
| 07:19.04 | SiNGLer | yes |
| 07:23.20 | KingDavidNYC | can somebody give me a pastebin site please |
| 07:23.24 | wdoekes2 | ~pb |
| 07:23.24 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 07:23.27 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 07:23.40 | wdoekes2 | and a good morning |
| 07:24.27 | KingDavidNYC | http://pastebin.us/2620 |
| 07:25.32 | SiNGLer | sipusers and sippeers replace asterisk with general |
| 07:26.15 | KNERD | fpaste is awesome...just type "fpaste <filename> and it returns a URL |
| 07:26.39 | KingDavidNYC | singler: sorry, I dont follow you |
| 07:27.01 | SiNGLer | sipusers => mysql,general,sip |
| 07:27.29 | KingDavidNYC | sinler: the name of the database is asterisk |
| 07:27.30 | SiNGLer | and same with sippeers |
| 07:27.46 | SiNGLer | just do it and ask questions later :) |
| 07:29.39 | KingDavidNYC | singler: asterisk is not complaining!! |
| 07:30.34 | KingDavidNYC | singler: great, but I still have the issue that it is not registering the phone when I use the realtime record |
| 07:31.02 | KingDavidNYC | singler: I have a record for username=100 which it does not register |
| 07:31.39 | SiNGLer | is there any other messages in log? is your table name "sip"? |
| 07:32.24 | KingDavidNYC | singler: yes, I have a table named sip with a record for username=100 |
| 07:32.30 | KingDavidNYC | this is what I get |
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| 07:33.38 | KingDavidNYC | handle_request_register: Registration from '"100"<sip:100@x.x.x.x>' failed for 'x.x.x.x' |
| 07:34.23 | KingDavidNYC | when I use disable realtime and use sip.conf, it registers |
| 07:34.45 | SiNGLer | can you pastebin your sip.conf definition and row in the table? |
| 07:34.49 | KingDavidNYC | which leads me to believe it is one of those freaking columns in the sip table |
| 07:40.03 | KingDavidNYC | this is the sp.conf that works: http://pastebin.us/2621 |
| 07:40.48 | KingDavidNYC | singler: excuse me, I dont know how to print the sip record vertically in mysql |
| 07:41.03 | SiNGLer | you can print it horizontally |
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| 07:42.34 | KingDavidNYC | ok, here it is: http://pastebin.us/2622 |
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| 07:45.03 | KingDavidNYC | singler: would it be what I am putting in xlite? |
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| 07:46.54 | SiNGLer | if password is specified correctly, then I don't know what is wrong, you can try to nullify md5secret, but I don't know if it interferes |
| 07:47.45 | KingDavidNYC | password is definitely correct |
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| 07:48.37 | KingDavidNYC | singler: what should I put in Authoriaztion user name in xlite? |
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| 07:49.13 | SiNGLer | 100 |
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| 07:50.08 | KingDavidNYC | xlite says: 405 forbideen, bad authorization name |
| 07:50.18 | KingDavidNYC | 403 |
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| 07:51.14 | SiNGLer | sorry, but I am out of suggestions... Maybe someone else will be able to help |
| 07:52.11 | KingDavidNYC | singler: interesting.. asterisk says wrong password1 |
| 07:52.25 | KingDavidNYC | singler: what do you mean nullify md5? |
| 07:52.42 | SiNGLer | did you nullified md5secret? |
| 07:52.47 | KingDavidNYC | no |
| 07:52.54 | SiNGLer | try doing so |
| 07:52.55 | KingDavidNYC | how do you do that? |
| 07:53.22 | KingDavidNYC | how? |
| 07:53.26 | SiNGLer | update sip SET md5secret = NULL |
| 07:53.32 | SiNGLer | wait |
| 07:53.40 | SiNGLer | I forget to add where name=100 |
| 07:53.41 | SiNGLer | :) |
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| 07:56.18 | KingDavidNYC | nada |
| 07:56.31 | KingDavidNYC | but it says "wrong password"! |
| 07:57.08 | SiNGLer | you reloaded sip module after modification? |
| 07:57.19 | KingDavidNYC | no |
| 07:57.34 | SiNGLer | do it then |
| 07:57.55 | KingDavidNYC | THAT DID IT!!!! |
| 07:58.06 | SiNGLer | :) |
| 07:58.32 | KingDavidNYC | It was the reload... I didn't know I had to do a reload on a realtime!!! |
| 07:59.01 | KingDavidNYC | SinGler: You are a genious man, thank you |
| 07:59.29 | SiNGLer | np |
| 07:59.37 | KingDavidNYC | singler: and guys, thanks...I am going to bed, it is 4am |
| 07:59.44 | SiNGLer | asterisk caches realtime |
| 07:59.57 | KingDavidNYC | bye bye, thank you |
| 08:00.10 | SiNGLer | good night KingDavidNYC |
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| 10:08.02 | iulhk | how can it possible, 2 sip peers already in call (sip:100 and sip:200 already in sip to sip call) if sip:300 will dial sip:100 or sip:200 he will get busy tone and disconneted instead of sending ring to sip:100 or sip:200 ? any idea? |
| 10:09.53 | SiNGLer | I didn't understand your question, if phone is busy with another call and that phone does not have actived call waiting feature, then the new call will get busy |
| 10:10.13 | kaldemar | iulhk: if the phones of asterisk are configured to accept only 1 call. |
| 10:11.13 | kaldemar | i'm tempted to ask how it could not be possible? |
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| 10:16.50 | c4rg | hi, did anyone used successfuly queues with context redirection when user presses key? |
| 10:20.12 | iulhk | <kaldemar> : i hv installed realtime asterisk, right now in field "call-limit" is by default "null", when user 100 and 200 already in sip to sip call and user 300 try to dial 100 then during call i am getting rings at user 100 client from user 300, i won't receive call from user 300 if i am already in call, user 300 should get busy tone and disconnected, if i will enable call-limit is 1 then |
| 10:20.12 | iulhk | would it help, bcoz somewhere i hv read that it will not help ? |
| 10:21.34 | kaldemar | iulhk: if the phone is configured to accept only 1 call, it doesn't matter what you do in asterisk. |
| 10:23.17 | kaldemar | iulhk: and call-limit was deprecated in 1.6.X already, you better not use it. |
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| 10:33.03 | kaldemar | c4rg: sure |
| 10:37.21 | iulhk | <kaldemar> using asterisk-1.6 i just entered call-limit=1 for all sip peers and now during call i am not getting 3rd user ring, my problem has been solved but according to u it's deprecated , what should i check now ?:( |
| 10:43.24 | c4rg | kaldemar: it looks like the call is hanged up (I can see that it goes to 'h' extension in the defined context; but it should go into the single digit extension) |
| 10:46.01 | kaldemar | what does the extension look like? |
| 10:46.43 | c4rg | something as simple as exten => 5,1,Verbose(test) |
| 10:47.08 | c4rg | if I remove this extension then pressing '5' doesn't give any effect |
| 10:47.21 | c4rg | so I guess asterisk knows about it |
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| 10:58.51 | c4rg | any ideas? |
| 11:06.30 | kaldemar | show a CLI output of a call with verbosity. |
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| 11:16.50 | c4rg | http://pastebin.com/9zhH8EP3 |
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| 11:31.39 | shapr | SHAZAM! |
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| 11:41.55 | c4rg | kaldemar: any ideas? ;) |
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| 11:46.57 | m_tadeu | hi...I'm having what it looks to be a nat problem. I've read all documentation asterisk/nat related and it looks like I have the proper configuration, but I still get no sound between 2 clients. The IVR sound reaches the caller properly, but I get no sound after an agent picks up the call |
| 11:47.35 | m_tadeu | asterisk is behind a nat and every client is behind some other nat |
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| 11:51.03 | m_tadeu | any ideas on how to solve this problem? |
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| 11:53.40 | kaldemar | c4rg: maybe the AGI is messing thing up for you. |
| 11:54.37 | kaldemar | m_tadeu: check your settings again. |
| 11:54.48 | kaldemar | m_tadeu: or let someone else take a look. |
| 11:55.49 | TeknoJuce | Hey Kaldemar Got it working with that patch |
| 11:56.23 | kaldemar | TeknoJuce: good to hear. |
| 11:56.24 | TeknoJuce | took me 4 hours to copy that patch manually to the latest unistim.c |
| 11:57.02 | TeknoJuce | google voice is now working with the nortel i2004 yay! |
| 11:57.09 | TeknoJuce | it so clear! |
| 11:57.22 | TeknoJuce | way better then the sip client on my computer (x-lite) |
| 11:58.26 | TeknoJuce | glad I kept on it, I was thinking this patch is never going to work after I am done with it haha |
| 11:59.40 | TeknoJuce | thanks for you help before kaldemar |
| 11:59.56 | kaldemar | no problem |
| 12:00.36 | TeknoJuce | do you know of an easier way to copy a .diff to a file where its has wildly changed since the diff was made? |
| 12:00.53 | TeknoJuce | as that was like pulling teeth :D |
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| 12:01.20 | hajekd | Hello, anyone using SS7 with ISDN failover switches? |
| 12:01.40 | tzafrir_laptop | ~nat |
| 12:01.40 | infobot | well, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 12:02.17 | kaldemar | TeknoJuce: if patch doesn't work, don't know of any automated way. having a diff viewer like meld open may help. |
| 12:02.48 | tzafrir_laptop | m_tadeu, generally: externip and localnet |
| 12:03.29 | tzafrir_laptop | Although some NAT routers support SIP in funny ways and thus mess it in creative ways |
| 12:03.46 | TeknoJuce | okay thanks any who I asked the guy that made the patch he responded the first time but not after that if he could remake the diff or add it to trunk think hes still fixing stuff though |
| 12:04.28 | TeknoJuce | I will have to test if the dialing in works from work tomorrow |
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| 12:04.49 | TeknoJuce | as I read there we're a few issues with that |
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| 12:15.12 | m_tadeu | kaldemar: here's a pastebin....I've triple checked already. http://pastebin.com/83yEcdN0 |
| 12:16.20 | m_tadeu | let me know if more info is needed |
| 12:20.22 | kaldemar | m_tadeu: you seem to have nat=no for m_tadeu. is that correct and on purpose? |
| 12:20.57 | m_tadeu | kaldemar: m_tadeu is inside the asterisk nat |
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| 12:22.44 | kaldemar | m_tadeu: what do you see in sip debug when you make a call? |
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| 12:34.41 | neurosys | burps |
| 12:36.01 | c4rg | kaldemar: yeah, that's possible. what should the agi do then? |
| 12:36.36 | c4rg | kaldemar: the agi is hanging up the channel (now) |
| 12:37.21 | kaldemar | c4rg: it shouldn't interfere with the dialplan flow. |
| 12:39.50 | c4rg | kaldemar: silently exit? |
| 12:40.46 | c4rg | kaldemar: wow, something started to work. thanks ;-) |
| 12:41.33 | m_tadeu | kaldemar: here's whar I see...http://pastebin.com/KjpkHbNd |
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| 12:53.53 | c4rg | hrm, is it possible to alter the way which dtmf keys are accepted inside queue application? |
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| 12:57.34 | m_tadeu | kaldemar: I see some "SIP/2.0 401 Unauthorized" on REGISTER requests(which I guess are keep-alive requests). Would this be a problem? |
| 13:00.57 | kaldemar | m_tadeu: no. unauthorized is a way to require authentication. using register requests for keep-alive is quite strange. it's possibly because of a short interval. |
| 13:02.03 | m_tadeu | kaldemar: should I use OPTIONS instead? my sip client allows this config |
| 13:02.44 | kaldemar | m_tadeu: your asterisk and the m_tadeu client seem to be running on the same host.. the client packets come from 82.154.x.x and SDP inside the packets have 192.168.1.10. since there is also nat=no for the device, that is a problem. |
| 13:03.02 | kaldemar | m_tadeu: OPTIONS is more common. |
| 13:04.39 | m_tadeu | kaldemar: indeed that ip addr is from the nat. I should set nat=yes then? |
| 13:06.49 | kaldemar | if asterisk sees the packets coming from the 82 address and your LAN is 192..., how are they in the same network? |
| 13:09.12 | m_tadeu | kaldemar: in deed we are. I changed the keep-alive to options and set the resgistrar to the asterisk internal ip. now m_tadeu is registered also with the internal ip address |
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| 13:19.16 | m_tadeu | kaldemar: new paste with the new settings....still no sound between client/agent. http://pastebin.com/Y5GECJ9k |
| 13:20.20 | *** join/#asterisk ssureshot (~digitolx@12.196.90.82) |
| 13:24.12 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 13:24.45 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 13:32.09 | ssureshot | does asterisk prebuild a web interface for the cdr lookup? |
| 13:32.10 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:34.54 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
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| 13:51.16 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:51.52 | *** join/#asterisk mawhii (~mawhii@123.219.119.70.cfl.res.rr.com) |
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| 14:11.11 | saliak | p3nguin: So, still having issues with the changes made yetserday. ultimately pared out all the extra stuff from extensions.conf (http://pastebin.com/5VuDrNq5) and sip.conf (http://pastebin.com/tccHyXJ7) and still behaves the exact same way. I am able to successfully register (it started working last night. i have a feeling my provider might have a thing where they temp disable registration after so many restarts in some amount of time), but |
| 14:11.12 | saliak | incoming calls still don't answer correctly (that is, i just get "the party you are trying to reach is not available to take your call..."). outgoing calls work great. |
| 14:17.22 | dandre | hello, |
| 14:17.48 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 14:18.32 | dandre | how can I have my extensions registered by users.conf using a gosub instead of macro-stdexten? |
| 14:21.28 | saliak | p3nguin: whoa! just started working! so the problem line seemed to be the "insecure" line. needed to be "insecure=port,invite". i thought i had tried that, but must have gotten lost in the permutations of other things i was trying |
| 14:23.54 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:28.55 | hajekd | Does anyone have experience with running 8xE1 in one server? |
| 14:29.12 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:29.13 | anonymouz666 | you mean two cards? |
| 14:29.15 | anonymouz666 | it works |
| 14:31.17 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 14:31.29 | *** join/#asterisk drcode (~user1@bzq-84-111-89-77.red.bezeqint.net) |
| 14:31.36 | drcode | hi all |
| 14:34.59 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 14:37.30 | hajekd | anonymouz666: Yes, two cards - probably TE420. Any issues with interrupts? |
| 14:38.11 | anonymouz666 | I didn't have any. |
| 14:38.31 | anonymouz666 | be sure you hardware is good enough to handle your calls. |
| 14:39.09 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:39.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:39.16 | drcode | can I use IAX or SIP with TCP port only? |
| 14:41.14 | WIMPy | Yes, my dual PIII-1266 couldn't handle the interrupts of a 4 plus a 2 port card. |
| 14:42.13 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 14:42.52 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:44.13 | hajekd | I'm thinking about some latest Dell PE server.... |
| 14:50.59 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 14:56.22 | leifmadsen | drcode: no, IAX2 uses UDP and 1.6.2 can use TCP for SIP |
| 14:58.36 | drcode | leifmadsen, how sip with tcp uses 1 port or more? |
| 14:59.19 | WIMPy | It's one port, but you still need UDP for RTP. |
| 14:59.28 | drcode | I see |
| 14:59.36 | drcode | iax uses 1 udp port? |
| 14:59.45 | WIMPy | yes |
| 14:59.50 | drcode | ok |
| 14:59.57 | drcode | thanx WIMPy |
| 14:59.58 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 15:13.46 | *** join/#asterisk torgnyw (~torgny.wa@25.79-161-149.customer.lyse.net) |
| 15:14.27 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:14.57 | davlefou | hi, |
| 15:15.15 | davlefou | our sip had been hacker |
| 15:15.24 | torgnyw | Hi, Anybody who has any experience with setting up Asterisk in passthrough mode with two PRI cards? Im planning on passing calls through to the old telephone system while implementing asterisk. |
| 15:15.56 | davlefou | We don't know how! Can you help me to find? |
| 15:16.20 | torgnyw | Then Im planning on adding some lines in extensions.conf for each phonenumber that's moved to the asterisk system, so Asterisk will handle the call instead of the old system |
| 15:16.54 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 15:17.08 | davlefou | I would like to find the way they have used! |
| 15:19.34 | *** join/#asterisk neurosys (~neurosys@209.50.97.18.nw.nuvox.net) |
| 15:21.23 | torgnyw | My plan is to write something like exten => _X.,1,Dial(${TRUNK_TO_OLD_SYSTEM}/{EXTEN}) from the context that the TRUNK from theTelephone Provider uses. But my problem is that I think the caller id will be Asterisk. Any one have any idea if this will work, and how to keep callerid? |
| 15:24.46 | pabelanger | davlefou: check the CDRs |
| 15:27.58 | drift- | :) |
| 15:28.19 | leifmadsen | has anyone heard of hearing "ringing" in the background of active calls? |
| 15:28.30 | leifmadsen | using a polycom ip335 with the latest firmware? (3.3.1) |
| 15:28.48 | leifmadsen | for some reason a couple of people complain (and I've heard it) that they hear ringing over top of their audio |
| 15:29.07 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:31.07 | leifmadsen | torgnyw: no, it won't be |
| 15:31.17 | leifmadsen | torgnyw: it'll be the callerID that came in |
| 15:32.37 | torgnyw | leifmadsen: are you sure, are there any settings needed in the dahdi config for this to work, or will it work by default? |
| 15:33.13 | *** join/#asterisk Cadey (~Cadey@62.84.178.106) |
| 15:33.22 | _Corey_ | leifmadsen: We saw that once where the call waiting was set to 'ringing' rather than the normal beep |
| 15:33.53 | *** join/#asterisk JunK-Y (~junky@pdpc/supporter/active/junk-y) |
| 15:34.03 | JunK-Y | yo! |
| 15:34.14 | Cadey | Hi guys, we want to play an out of hours message and then have the caller land in a mailbox so they can leave a message. How can I supppress the comedian mail message saying the person at Blah is away/unavailable and have it simply say leave a message then a beep? |
| 15:34.20 | davlefou | pabelanger: CDRs? |
| 15:34.35 | JunK-Y | when using, Dial(${TRUNK}/5145551234,,M(page-pre-audio)D(#7)) , the Macro is called before sending the dtmfs, is there any way to specify the order without touching app_dial.c ? |
| 15:35.21 | pabelanger | leifmadsen: Do you have any DAHDI channels involved? |
| 15:35.34 | pabelanger | davlefou: Call Detail Records |
| 15:35.47 | thehar | hops around |
| 15:35.50 | davlefou | ok |
| 15:36.09 | WIMPy | davlefou: Nothing special to that. Just keep your Dialplan (almost) empty. |
| 15:36.26 | elb | leifmadsen: FYI, I haev the same problem Lakesidetech has here: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 15:37.13 | elb | leifmadsen: oh, my mistake, I thought you were one of the commenters who could not reproduce, I misremembered |
| 15:37.21 | Cadey | oh I thikn i see |
| 15:37.23 | davlefou | WIMPy, pabelanger An special file? |
| 15:37.38 | pabelanger | JunK-Y: Can you not use SendDTMF() within your macro? |
| 15:38.13 | davlefou | or an special keyword? |
| 15:38.38 | pabelanger | FWIW: I never new about the D option for dial :) I assume it is used for dialing long distance access codes? |
| 15:38.58 | WIMPy | davlefou: Dialplan = extension.conf |
| 15:40.16 | davlefou | WIMPy: Why? |
| 15:40.25 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1279411008.dsl.bell.ca) |
| 15:40.29 | WIMPy | Why what? |
| 15:40.53 | *** join/#asterisk Tim_Toady (~moi@77.49.3.200) |
| 15:42.07 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:42.21 | davlefou | WIMPy: I don't undustand aout extention.conf |
| 15:42.33 | davlefou | WIMPy: I don't undustand about extention.conf |
| 15:43.30 | JunK-Y | pabelanger: yeah ,good idea. |
| 15:43.36 | WIMPy | If you only want pass-thru you only need one context per port, each only containing a Dial() to the other port. The rest is in the book. |
| 15:43.45 | WIMPy | ~newbook |
| 15:43.46 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 15:45.01 | *** join/#asterisk dlynes_laptop (~dlynes@bas6-hamilton14-1176141513.dsl.bell.ca) |
| 15:45.27 | davlefou | I try to find if our server has been hacked or not, because, it can't become from my sip only! |
| 15:46.53 | JunK-Y | pabelanger: i would have trouble if i used D(:#7), but D(#7) is fine. thanks for suggestion. |
| 15:46.54 | davlefou | How can we have un clear answer? |
| 15:47.12 | WIMPy | davlefou: Sorry. Mixed up the lines. That wasn't about you. |
| 15:47.29 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 15:47.32 | davlefou | WIMPy: ok, can you help me? |
| 15:48.48 | WIMPy | davlefou: Check your contexts and if you have nat support, check that you have enabled strictrtp in rtp.conf. |
| 15:50.12 | davlefou | <PROTECTED> |
| 15:52.38 | leifmadsen | _Corey_: aha thanks, let me see if that could be an issue |
| 15:53.46 | davlefou | WIMPy: Is it ok? |
| 15:54.24 | WIMPy | davlefou: It's commented out. That makes a no. |
| 15:55.04 | davlefou | ok |
| 15:55.40 | *** join/#asterisk Deeewayne (~dwayne@68-64-220-110.static.forethought.net) |
| 15:55.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:55.44 | leifmadsen | _Corey_: hmmm was worth a shot, but.... <call.callWaiting call.callWaiting.ring="beep"> |
| 15:55.54 | davlefou | WIMPy: Sorry, it y already commented! |
| 16:00.32 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
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| 16:05.53 | nestAr | anyone using Vitelity's SMS service with asterisk? |
| 16:08.15 | mersault | is it possible to fire a call straight into google voice's voicemail? I have a regular SIP account and DID from another provider already, in addition to my google voice DID. I'd like to integrate the voicemail functionality for them both onto google voice. |
| 16:09.16 | kaldemar | davlefou: what version of asterisk are you using? |
| 16:09.30 | *** join/#asterisk fordfrog (~fordfrog@gentoo/developer/fordfrog) |
| 16:09.46 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 16:10.44 | fordfrog | hi, i have issue with ael and WaitExten(), i get timeout all the time and the pressed exten is not caught ... i use asterisk-1.6.2.16.2 ... anyone has an idea what i might do wrong? |
| 16:11.19 | m_tadeu | http://pastebin.com/CWpbZafk |
| 16:12.11 | _Corey_ | leifmadsen: Only thing I could think of... it's probably something ugly |
| 16:12.34 | kaldemar | m_tadeu: do you see rtp packets going to both clients with "rtp set debug on"? |
| 16:13.35 | fordfrog | this is the part of the dial plan that should handle it: http://pastebin.com/fB5zY93M |
| 16:13.58 | m_tadeu | kaldemar: good point....no packets when the agent answers the call |
| 16:15.53 | *** join/#asterisk Freeaqingme_ (~dolf@dsl-083-247-011-232.solcon.nl) |
| 16:16.44 | davlefou | kaldemar: 1:1.6.2.5 |
| 16:18.56 | *** join/#asterisk heliosj (jeff@pdpc/supporter/student/xheliox) |
| 16:19.12 | kaldemar | davlefou: http://www.asterisk.org/security <-- your asterisk was released in 2010-02-25, so a bunch of those vulnerabilities apply. |
| 16:19.24 | *** part/#asterisk heliosj (jeff@pdpc/supporter/student/xheliox) |
| 16:21.16 | Cadey | any english peeps in there that has the sound likes for asterisk that say "hash" and not "pound" for the # key ? |
| 16:21.37 | Cadey | I have english voice overs on the system now but even they still say pound :( |
| 16:28.47 | fordfrog | i just upgraded to asterisk-1.6.2.17 but WaitExten() does not work there either :-/ |
| 16:29.47 | leifmadsen | _Corey_: ya, I have a feeling it has something to do with the call forwarding on the lines.... |
| 16:33.26 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 16:35.05 | neurosys | Hmm Im trying to debug an issue. I have registration, NP. But when an actual call takes place, its not getting the ACK packet back. Any thoughts? |
| 16:35.20 | Freeaqingme_ | firewall? |
| 16:35.25 | neurosys | ASA |
| 16:35.48 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 16:35.58 | m_tadeu | I'm googling for rtp problems, but I can't seem to find good info...can anyone direct me for some info on why there's no rtp packets? |
| 16:36.17 | Freeaqingme_ | neurosys, I meant, maybe your firewall is the culprit? |
| 16:36.51 | kaldemar | neurosys: what is not getting an ACK? |
| 16:36.51 | neurosys | Freeaqingme: heh indeed. I have done ASA's before without a problem. And why would the reg get the ack via sip but not the call? |
| 16:37.07 | neurosys | kaldemar: the call buildup. |
| 16:37.32 | neurosys | kaldemar: call starts.. I get RTP... but of course after the 10th try it drops the call |
| 16:37.44 | Freeaqingme_ | neurosys, maybe you're using both tcp & UDP? |
| 16:38.08 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 16:38.24 | fordfrog | m_tadeu: your call is in the same network (no firewall)? |
| 16:38.38 | kaldemar | neurosys: "what" as in asterisk or the peer... |
| 16:39.04 | m_tadeu | fordfrog: I have a client outside the asterisk nat and an agent inside |
| 16:39.05 | kaldemar | neurosys: but if you have retransmissions, it sounds like a firewall issue at the client end. |
| 16:39.45 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 16:40.08 | fordfrog | m_tadeu: that's what i've just set up, just can't get waitexten work ... so it does not work for you only when the call is made from outside, making the call from inside works? |
| 16:40.36 | *** join/#asterisk fauxalliance (~fauxallia@142.162.116.237) |
| 16:41.49 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85de.bcn.adamo.es) |
| 16:42.05 | neurosys | kaldemar: but wouldnt a FW issue effect the reg as well? |
| 16:42.24 | *** join/#asterisk Alric (~alric@64.6.54.218) |
| 16:42.28 | neurosys | kaldemar: oh... maybe not... |
| 16:42.38 | neurosys | kaldemar: since Im generating the request thru the FW |
| 16:43.09 | neurosys | kaldemar: but yet, the SIP request to the * box gets there, the call starts and goes to ack, but doesnt get the ack back. |
| 16:43.49 | m_tadeu | fordfrog: doesn't seem to work either |
| 16:43.52 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan.noare-1.holmedal.net) |
| 16:43.53 | kaldemar | neurosys: yes. the firewall may drop the hole to your client during inactivity. use the qualify settings for keep-alive. |
| 16:44.23 | fordfrog | m_tadeu: and when running asterisk in debug and verbose mode, it does not complain about anything? |
| 16:45.16 | neurosys | kaldemar: qualify is on. again tho, why would reg work, its set to 120 sec.. and the call makes it to the * box, RTP starts... but when looking for the ack back from the provider... it's not getting it :/ |
| 16:45.23 | kaldemar | neurosys: or there may be an ALG in the router that screws things up. anyway, try to get a sip debug of a failed call, it will help debugging the issue. |
| 16:45.45 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 16:45.46 | m_tadeu | fordfrog: it doesn't |
| 16:46.06 | neurosys | kaldemar: the ASA does have an ALG (policy mappings and inspection). It is enabled. |
| 16:46.21 | *** join/#asterisk cerberus_za (~coert@196-215-125-162.dynamic.isadsl.co.za) |
| 16:46.24 | fordfrog | m_tadeu: so you set core set debug 5 and core set verbose 5 and nothing? |
| 16:47.29 | kaldemar | neurosys: try without the ALG, if it indeed is a SIP ALG. it will most likely just cause problems. |
| 16:47.42 | fordfrog | m_tadeu: i had issue here that phone was using g729 codec which is not supported on asterisk so when somebody called in, nothing could be heard |
| 16:48.20 | neurosys | kaldemar: tryuing.. |
| 16:50.02 | m_tadeu | fordfrog: how can I check which protocol is being used? |
| 16:51.00 | fordfrog | m_tadeu: proly 'sip show channels' should reveal it during call |
| 16:51.29 | m_tadeu | fordfrog: it's using ilibc |
| 16:52.21 | m_tadeu | but, what can originate that no rtp packets are moving around? |
| 16:53.04 | fordfrog | if the call does not work even locally, then it's not firewall related |
| 16:54.37 | *** join/#asterisk pushpop- (~pushpop@pool-173-77-243-69.nycmny.fios.verizon.net) |
| 16:54.37 | fordfrog | m_tadeu: ilibc or ilbc? |
| 16:55.15 | pushpop- | hi all, I'm having an issue where my default unavailable greeting will not play in the cli i get no errors on the phone line it rings then goes silent when the voicemail picks up |
| 16:55.17 | m_tadeu | fordfrog: sorry...ilbc..bad habit |
| 16:55.18 | pushpop- | any idea's? |
| 16:56.18 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 16:56.25 | fordfrog | pushpop-: turn on debug :-) |
| 16:56.36 | fordfrog | m_tadeu: so both sides are using ilbc? |
| 16:56.37 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 16:56.45 | m_tadeu | fordfrog: yes |
| 16:56.47 | pushpop- | frodfrog its ok |
| 16:57.36 | p3nguin | saliak: Yeah... |
| 16:57.39 | p3nguin | (2043.29) <p3nguin> insecure=very should be insecure=port,invite |
| 16:57.51 | fordfrog | m_tadeu: i have no experience with ilbc to be honest, using alaw, ulaw and gsm ... you can't make any call locally on your asterisk or just to the agent? |
| 16:59.56 | m_tadeu | fordfrog: if I call directly(without the IVR in between) the same happens |
| 16:59.56 | p3nguin | crap |
| 17:00.24 | p3nguin | He quit before I could tell him what else is wrong with his conf. |
| 17:01.02 | pushpop- | fordfrog, any other idea's |
| 17:01.13 | pushpop- | <PROTECTED> |
| 17:01.52 | *** join/#asterisk lost_soul (shawn@cpe-74-78-191-114.twcny.res.rr.com) |
| 17:01.52 | fordfrog | m_tadeu: then maybe your nat in sip is not cofigured correctly ... might be? |
| 17:02.27 | fordfrog | pushpop-: without any debug error i have no idea, sorry |
| 17:02.50 | pushpop- | ok thank you |
| 17:02.50 | m_tadeu | fordfrog: allow me to recheck...please confirm that I only need to redirect on the asterisk nat |
| 17:04.00 | pushpop- | fordfrog, http://pastebin.com/NkeXp5M3 here is cli w/ debuggin when voicemail kicks in |
| 17:04.12 | pushpop- | its completely silent on the phone when the cli voicemail picks up |
| 17:05.27 | fordfrog | pushpop-: so none of the 4 sounds that are in log play? |
| 17:05.33 | pushpop- | correct |
| 17:06.56 | fordfrog | m_tadeu: check this, this works for me: http://pastebin.com/Kcw20VXn |
| 17:07.32 | fordfrog | m_tadeu: both local calls and external calls work, though external calls are also firewall configuration dependent |
| 17:10.32 | citywok | I've got one phone number that I have problems calling. Looking at the pcap of a call to that number i only get a 183 Session Progress after the 100 Trying, but no 180 Ringing message. |
| 17:11.00 | citywok | It typically takes 32 seconds for that number to answer, and when it does (it's a conference bridge) it immediately says we're sorry we didn't understand the PIN you provided. thoughts? |
| 17:11.12 | m_tadeu | fordfrog: what is preferable to use? externip or externhost? |
| 17:11.13 | citywok | this happens on 3 seperate SIP carriers (bandwidth.com, flowroute, and qwest) |
| 17:12.29 | fordfrog | m_tadeu: no idea, i made this setup long time ago, just had to adjust it for 1.6 when i upgraded it several months ago |
| 17:13.12 | pushpop- | fordfrog, no ideas for me =P |
| 17:13.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 17:14.22 | fordfrog | pushpop-: idk, maybe similar issue as m_tadeu? it plays the sound, but you can't hear it, so maybe rtp does not come throught? |
| 17:14.27 | *** join/#asterisk Freeaqingme_ (~dolf@dsl-083-247-011-232.solcon.nl) |
| 17:14.28 | fordfrog | through* |
| 17:14.48 | pushpop- | rtp? |
| 17:15.38 | citywok | rtp is the media stream of the call |
| 17:16.25 | pushpop- | is there a way to tell if that is workign correctly? |
| 17:16.56 | citywok | use tcpdump to make a dump, and open it in wireshark which will be able to show you the full call path for sip, and play the actual call audio |
| 17:17.43 | pushpop- | ok |
| 17:19.02 | fordfrog | anyone has idea why in this script, waitexten does not catch the pressed numbers and i get timeout instead? http://pastebin.com/fB5zY93M |
| 17:22.05 | *** join/#asterisk m_tadeu (~quassel@static-b5-252-50.telepac.pt) |
| 17:25.22 | citywok | fordfrog: do you know that DTMF is working properly? |
| 17:26.45 | fordfrog | citywok: that i unfortunately have no idea, whether it works or what i should set where, i tried to read something about it but found nothing useful |
| 17:27.02 | citywok | call 1800COMCAST and see if it will let you enter a phone number. lol |
| 17:28.09 | fordfrog | would not be cheap here from czech rep i guess :-P |
| 17:33.12 | *** join/#asterisk sahafeez (~sahafeez@4.53.128.211) |
| 17:35.05 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:38.12 | m_tadeu | even setting asterisk on the dmz wont solve the problem |
| 17:38.45 | Freeaqingme_ | fordfrog, there are also czech ivr/aa phonenumbers where you can enter some digits |
| 17:38.45 | p3nguin | Most people have no idea how to use DMZ or even what it really does. |
| 17:39.15 | fordfrog | m_tadeu: are you sure there is no firewall blocking rtp on your asterisk machine? |
| 17:39.16 | *** join/#asterisk mawhii (~mawhii@123.219.119.70.cfl.res.rr.com) |
| 17:40.09 | m_tadeu | fordfrog: I just set the asterisk host on the dmz...the host itself has no firewall |
| 17:40.57 | fordfrog | Freeaqingme_: well, there should be no issue with dtmf from the cell phone or hw sip phone generally, but for some reason, dtmf is not detected at all when calling to asterisk ... i also checked dtmf settings for sip and it's the default one |
| 17:41.35 | Freeaqingme_ | fordfrog, also on your phone? |
| 17:41.52 | Freeaqingme_ | I have been debugging this in the past for hours, only to find out my dtmf settings on the phone itself werewrong |
| 17:43.23 | fordfrog | Freeaqingme_: well, client calls from cell phone to provider which routes the call to asterisk, where ivr is set up, but asterisk does not "hear" dtmf signals at all ... it waits and then issues timeout |
| 17:44.16 | *** join/#asterisk [netman] (~netman@3.Red-83-36-36.dynamicIP.rima-tde.net) |
| 17:44.21 | *** join/#asterisk GTXComm (~John@cpe-72-128-62-30.kc.res.rr.com) |
| 17:44.34 | Alric | If I have a queue set "joinempty => unavailable,inuse,ringing" and all the static members in it are marked Busy, shouldn't a call into the queue not join the queue, but go on to the next dialplan priority? |
| 17:44.36 | fordfrog | m_tadeu: i think there should be some rtp info in the sip packets when negotiating the connection, did you check whether it looks ok? |
| 17:44.38 | Freeaqingme_ | dunno about that. My upstream provider has some good dsp's that interpret the dtmf tones for me and convert them to plain sip |
| 17:45.17 | GTXComm | Hello Asterisk. |
| 17:45.40 | fordfrog | Freeaqingme_: hm, maybe i should try to change the settings and see if any non-default works then |
| 17:50.13 | m_tadeu | why is it recommended such a big range for rtp(10000-20000)? |
| 17:53.27 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 17:54.51 | psilikon | m_tadeu, http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf I was actuallt just checking that out |
| 17:55.08 | psilikon | each connection can use as many as four ports |
| 17:56.52 | m_tadeu | I see....but I hardly use 2500 simultaneous calls |
| 17:58.39 | Freeaqingme_ | If I set a grandstream gxp2120 on 'busy', does anybody know what that actually does? |
| 17:59.05 | psilikon | m_tadeu, really? I use 5000. All the cool ppl use atleast 3000 |
| 17:59.17 | nestAr | lol |
| 17:59.31 | psilikon | m_tadeu, yeah me neither. I guess it is there if you need it. |
| 18:00.08 | m_tadeu | psilikon: hehe I'm not that cool yet |
| 18:02.13 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 18:02.27 | psilikon | m_tadeu, you'll get there one day my friend. |
| 18:03.14 | p3nguin | saliak: As I've tried to mention a couple of times, now... |
| 18:03.21 | p3nguin | (2043.29) <p3nguin> insecure=very should be insecure=port,invite |
| 18:04.09 | p3nguin | saliak: I also noticed that lines 18-23 of your sip.conf paste belong ABOVE the register statement. You've got them in a peer definition for some unknown reason. |
| 18:04.20 | *** join/#asterisk [netman] (~netman@181.Red-80-39-202.dynamicIP.rima-tde.net) |
| 18:04.25 | Alric | Anyone familiar with the queue option "joinempty"? I'm having trouble getting it to keep calls out of an empty queue. |
| 18:08.03 | saliak | p3nguin : you are correct my friend. interesting. that was from the example broadvoice gave me. |
| 18:09.35 | *** join/#asterisk m_tadeu (~quassel@static-b5-252-50.telepac.pt) |
| 18:10.29 | m_tadeu | still no rtp packets moving around.... |
| 18:10.43 | p3nguin | saliak: They told you that it needs to be insecure=very, or they told you to put general options within the peer entry? |
| 18:10.57 | saliak | p3nguin: both |
| 18:11.27 | saliak | p3nguin: there were some other mistakes that I caught on my own as well, but they're clearly quite sloppy about things |
| 18:11.30 | p3nguin | saliak: I'm quite certain that insecure=very used to be valid on previous Asterisk versions. |
| 18:11.56 | p3nguin | bindaddr, on the other hand, has never been valid within the peer definition. |
| 18:12.16 | saliak | p3nguin: yeah, that's what I was able to tell from some web searching. i think they just haven't updated their stuff in a very long time. that's interesting. maybe it just happened to work? |
| 18:13.08 | p3nguin | My guess is that they pulled some crap off the web and compiled a "sample" to give to their customers. They know absolutely nothing about configuring Asterisk. |
| 18:13.43 | saliak | p3nguin - very likely. |
| 18:15.48 | *** join/#asterisk [netman] (~netman@33.Red-79-153-220.dynamicIP.rima-tde.net) |
| 18:16.18 | *** join/#asterisk imox1234 (~imox1234@p4FC5C779.dip0.t-ipconnect.de) |
| 18:16.52 | m_tadeu | what can I check if there's no rtp packets moving around? I already checked the firewall, which looks fine |
| 18:18.32 | Freeaqingme_ | m_tadeu, tried something like ethereal/wireshark? |
| 18:20.25 | m_tadeu | Freeaqingme_: not really...but asterisk should get rtp packets from the internal network, I guess |
| 18:21.10 | Freeaqingme_ | yeah, but if you run wireshark on both locations, you can see if they at least arrive, or if there's more info |
| 18:25.45 | *** join/#asterisk voxter (~voxter@207.216.70.109) |
| 18:25.47 | thehar | ~/win 27 |
| 18:25.49 | thehar | ugh |
| 18:26.01 | m_tadeu | Freeaqingme_: I'll check it out....on one end I'm only getting SIP and RTCP packets...gonna check the other end |
| 18:26.27 | *** join/#asterisk ssureshot (~digitolx@12.196.90.82) |
| 18:26.47 | ssureshot | what are you guys using for cdr record lookup? |
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| 18:30.19 | retentiveboy | Anybody working on a way for inbound Jabber messages to be handled with dialplan logic? |
| 18:30.53 | retentiveboy | Jabber messages not associated with a current call that is. |
| 18:31.47 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:32.31 | Freeaqingme_ | retentiveboy, perhaps the dev channel is a better place to ask |
| 18:33.30 | retentiveboy | Freeaqingme_: perhaps |
| 18:35.30 | m_tadeu | Freeaqingme_: no RTP packets are getting to/reaching in both ends |
| 18:35.57 | Freeaqingme_ | Okay, so what is the problem then m_tadeu ? |
| 18:36.05 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
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| 18:38.33 | m_tadeu | Freeaqingme_: well I kinda know that....just don't know how to solve it. why aren't those packets moving around? btw, when the call is answered by the IVR the RTP packets flow as expected |
| 18:41.10 | *** join/#asterisk AMindMobile (~kvirc@95-28-106-82.broadband.corbina.ru) |
| 18:42.22 | Freeaqingme_ | m_tadeu, I'm sorry, Im afraid I cant answer that |
| 18:43.00 | saliak | p3nguin: so i'm trying to manually set the caller id of my outgoing calls (http://pastebin.com/39vB95yh) with Set(CALLERID(name)="asdf") before I do a Dial command. is there anything else I need to do? for some reason this isn't reflected on the device I get the call on |
| 18:44.02 | m_tadeu | Freeaqingme_: well, thanx for your pacience, anyway :) |
| 18:46.13 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 18:52.09 | p3nguin | saliak: Set(CALLERID(num)=4015621302) |
| 18:52.15 | m_tadeu | confirm me something....rtp packets always goes through asterisk or can I set it only between sip phones? |
| 18:52.16 | p3nguin | saliak: You can't set the name. |
| 18:52.28 | p3nguin | saliak: Well, you can, but it won't do any good. |
| 18:53.14 | *** join/#asterisk [netman] (~netman@200.Red-83-36-38.dynamicIP.rima-tde.net) |
| 18:53.16 | saliak | p3nguin: ahh. ok . it seems to be an option. does it only do something for internal transfers, etc? or calls that don't go out to a trunk? |
| 18:54.33 | p3nguin | saliak: Caller ID name (known as CNAM) on the PSTN is looked up in a database by the receiving telco. So no matter what CALLERID(name) you set, it's worthless. |
| 18:56.29 | saliak | p3nguin: cool, makes sense. Is there a chance that setting the number won't do anything either (shows my 401 number, if I set it or not)? I'm trying to setup asterisk to forward calls to our cell phone,but send some identifying feature so we know which extension was selected (so that could be either the name, which is impossible, or the CID number, that's unique) |
| 18:58.30 | p3nguin | saliak: It depends on your ITSP if they will allow you to set your own CID number. I'll give you two options to set for the peer that you're sending calls outbound through... and then you can try setting your CID number to some other 10-digit number. Just one moment. |
| 18:58.49 | p3nguin | trustrpid=yes |
| 18:58.49 | p3nguin | sendrpid=yes |
| 18:59.03 | p3nguin | Add those to your broadvoice peer. |
| 18:59.16 | *** join/#asterisk nighty^ (~nighty@tin51-1-82-226-147-104.fbx.proxad.net) |
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| 19:01.07 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 19:01.18 | saliak | p3nguin: hrm. doesn't seem to change anything. can you recommend a ITSP that lets you set the CID number? |
| 19:02.01 | p3nguin | saliak: VoIP.ms and Flowroute both allow it. |
| 19:04.35 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:12.16 | psilikon | voip.ms has been great in my experience |
| 19:12.54 | psilikon | Vitelity was the best experience but was probably due to my lack of experience working with Vicidial |
| 19:13.17 | saliak | psillikon: is the support good? |
| 19:13.28 | nestAr | I've used Vitelity for a while now, never had a problem. Support is reasonably quick to reply. |
| 19:15.17 | p3nguin | VoIP.ms is a reseller for Vitelity, and they have cheaper pricing. |
| 19:15.46 | *** join/#asterisk imox1234 (~imox1234@p4FC5C779.dip0.t-ipconnect.de) |
| 19:16.38 | *** join/#asterisk imox1234 (~imox1234@p4FC5C779.dip0.t-ipconnect.de) |
| 19:19.33 | mersault | anyone know of a way to send a call directly to voicemail at google voice without using do not disturb? I'd like to use google voice as a shared voicemail system. |
| 19:20.20 | *** join/#asterisk hajekd (~textual@2a00:1268:15:1800:fa1e:dfff:fee0:e515) |
| 19:21.43 | p3nguin | mersault: Do you still want to be able to answer phone calls as well? |
| 19:23.13 | mersault | yeah. my thinking is that I have a couple of other DIDs from regular SIP providers |
| 19:23.40 | mersault | but if I could shunt a call directly into the voicemail at google, then I could use it for my voicemail instead of asterisk voicemail |
| 19:23.52 | p3nguin | mersault: If you didn't want to take calls as well, remove all the forwarding phone numbers, and voicemail should be the only thing left to take a call to your gv number. |
| 19:24.02 | mersault | this guy: http://blog.hoopycat.com/2010/07/voicemail-notifications-with-asterisk-and-google-voice seems to have gotten pretty close |
| 19:24.22 | mersault | p3nguin: problem is, I also want to keep and use my google voice DID |
| 19:26.01 | mersault | also, I'm in Canada, and google won't let me re-add my canadian cell phone number to my account as a forwarding number if I delete it |
| 19:26.08 | mersault | so it would become inflexible in the future. |
| 19:26.13 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 19:27.38 | p3nguin | You can untick any forwarding number that's on the account -- no need to delete. |
| 19:28.32 | mersault | oh, and if I send a call to my GV DID over the google voice channel, it lands at the internal voicemail prompt |
| 19:28.34 | mersault | that also sucks |
| 19:37.56 | mersault | hmmm... alternatively, is there any way to get asterisk to answer a call to my GV DID? |
| 19:38.28 | Qwell | ~google voice |
| 19:38.29 | mzb | [Google] http://webcache.googleusercontent.com/search?q=cache:wWv1HfDklwoJ:https://www.google.com/voice+voice&cd=1&hl=en&ct=clnk&source=www.google.com [Cached] |
| 19:38.37 | Qwell | mzb: Please turn that off. |
| 19:39.00 | Qwell | mersault: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 19:40.15 | mersault | Qwell: hmm.. I have that setup mostly the same, but I didn't see anything in the asterisk cli when I tried calling my DID. I'll keep fiddling. probably just fat fingered something |
| 19:42.07 | *** join/#asterisk cyphorious (~cyphoriou@212-183-79-247.adsl.highway.telekom.at) |
| 19:42.38 | p3nguin | If you don't get calls directly to Asterisk, you probably didn't untick all forwarding numbers and tick the google chat box in the settings for your gv account. |
| 19:43.15 | *** part/#asterisk Alric (~alric@64.6.54.218) |
| 19:44.29 | mersault | grrr... outbound calls are fine, but inbound are not appearing in asterisk |
| 19:44.47 | mersault | the only phone ticked is google chat |
| 19:47.40 | mersault | oh, and what is a 'guest call' in the context of gtalk? |
| 19:48.47 | elb | any incoming call from a jid that's not in your buddy list and doesn't have its own stanza in gtalk.conf |
| 19:49.13 | mersault | would that happen to include calls to my GV DID? |
| 19:49.33 | mersault | would I otherwise have to create a stanza for my DID? |
| 19:51.34 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
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| 20:02.17 | mersault | aha... looks like it's related to jabber priority. |
| 20:02.33 | mersault | if I closed gmail and google voice tabs, the next call hit my asterisk box. |
| 20:02.52 | Qwell | weird. it should have gone to all, I'd think |
| 20:02.55 | retentiveboy | mersault: use a GV account dedicated to the PBX so calls don't go to your browser or chat client. |
| 20:03.03 | Qwell | and yeah, that |
| 20:03.21 | mersault | problem is the DID is already associated with my primary gtalk ID |
| 20:03.33 | Qwell | disassociate it, or get a new acct |
| 20:03.56 | Qwell | You can move a DID to another GV account, if I'm not mistaken. |
| 20:04.01 | retentiveboy | Think that's spelled out in the docs pretty clearly. Bit me too though :) |
| 20:04.26 | mersault | interesting... 'cause that would solve my problems quite nicely. |
| 20:05.12 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 20:05.27 | mersault | ah damn, but that's gonna cost me 10 bucks... |
| 20:05.40 | Qwell | 10 whole dollars? |
| 20:07.27 | mersault | it's not the 10 bucks, it's that I didn't pay for the original DID |
| 20:08.27 | mersault | also, this is purely a toy for me. |
| 20:08.58 | Qwell | $10 makes for a pretty cheap toy |
| 20:09.13 | mersault | but still 10 dollars more than a free toy. |
| 20:09.15 | Qwell | that's what, 2 (bad) beers? |
| 20:09.44 | outtolunc | Add a couple zeros to that for my latest toy |
| 20:09.59 | mersault | also, since the Canadian support is completely missing, paying for it seems a little silly. |
| 20:10.43 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
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| 20:13.11 | Qwell | outtolunc: yeah, I hear you there. Mine was about 2000x that. |
| 20:13.19 | mersault | as soon as they will let me enter a canadian forwarding number, or pair with Rogers like they have sprint, I'll jump at google voice |
| 20:13.31 | mersault | until then, it's just something to amuse my when I'm bored at work. |
| 20:13.44 | mersault | and I'll take my toy money for something more interesting |
| 20:14.59 | Qwell | I recommend a gocart. |
| 20:15.23 | mersault | that actually would be a ton of fun. |
| 20:16.44 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 20:20.09 | mersault | oh, any decent user portals for asterisk? I only really care about call logs and voicemail. freebsd is overkill |
| 20:25.05 | Qwell | I think you mean FreePBX.. |
| 20:25.15 | Qwell | FreeBSD would indeed be overkill for that. |
| 20:25.25 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 20:26.52 | mersault | Qwell: you would be correct. freebsd is muscle memory at this point. I did mean freepbx. |
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| 20:33.04 | citywok | does anybody else use bandwidth.com and just have problems for the last hour? |
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| 21:23.24 | citywok | oh man i feel bad for anybody that uses bandwidth.com as their primary carrier liek i do. hopefully you have a few backup sip accounts that work. |
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| 21:32.01 | drift- | voip.ms and didforsale.com ftw! |
| 21:36.14 | leifmadsen | citywok: odd, because I never have problems with them |
| 21:36.31 | leifmadsen | perhaps different types of service? |
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| 22:23.02 | khaluk | hi all |
| 22:23.16 | khaluk | I have a problem with my elastix. |
| 22:23.57 | khaluk | I am using TDM410 card and it doesnt detect hangup on FXO . Is there any body can talk about it? |
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| 22:34.27 | khaluk | is there anybody here from TURKEY? |
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| 22:54.02 | Serees | hi, anybody here... I have some issues with the gentoo install of asterisk using DAHDI and a HFC BRI Card |
| 22:54.25 | Serees | it seems as if asterisk is not recognizing my signalining... and vise versa |
| 22:54.31 | Serees | anybody can help on this? |
| 22:55.11 | WIMPy | Describe what you try to do and what happend (or dosn't happen). |
| 22:55.59 | tzafrir_laptop | Serees, do you see in in dahdi_hardware? |
| 22:56.04 | tzafrir_laptop | lsdahdi? |
| 22:57.55 | Serees | ### Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] " (MASTER) AMI/CCS |
| 22:57.55 | Serees | <PROTECTED> |
| 22:57.55 | Serees | <PROTECTED> |
| 22:57.55 | Serees | <PROTECTED> |
| 22:57.55 | Serees | ### Span 2: WCFXO/0 "Wildcard X100P Board 1" RED |
| 22:57.55 | Serees | <PROTECTED> |
| 22:57.55 | Serees | my signaling line used to be signalling = bri_cpe_ptmp |
| 22:58.05 | Serees | this worked fine until I upgraded |
| 22:58.21 | Serees | then asterisk saw the call comming in |
| 22:58.36 | Serees | dail plan was executing ok, but the line kept on ringing |
| 22:59.01 | Serees | the 'answer' command did not seem to go through to the line |
| 22:59.01 | WIMPy | When Asterisk sees the call, the signalling can't be that wrong. |
| 22:59.34 | Serees | ok... then why is the line not picked up? |
| 22:59.58 | Serees | I read somewhere that with DAHDI the signlaing should be bir_cpe_ptmp |
| 22:59.59 | WIMPy | You can try 'pri set debug 2 span 1' |
| 23:00.11 | Serees | ok, sec |
| 23:00.19 | WIMPy | Do you use current versions of libpri and Asterisk? |
| 23:01.01 | Serees | libpri I have no idea... asterisk is 1.4 something (latest one from gentoo tree) |
| 23:01.12 | fauxalliance | latest my ass |
| 23:01.35 | WIMPy | Uh. When using BRI you should really use recent versions. |
| 23:01.47 | kerframil | Serees: asterisk-1.4 isn't in the portage tree anymore so it can't be the latest. I know, becuase I have a 1.4.40 ebuild in my own overlay ;) |
| 23:01.56 | fauxalliance | like.. at least 1.6... keep current with the documentation |
| 23:02.21 | WIMPy | I'd very much suggest 1.8. |
| 23:02.26 | Serees | Connected to Asterisk 1.6.2.16.2 currently running on pabx (pid = 23787) |
| 23:02.30 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 23:02.52 | kerframil | Serees: if you can pinpoint the regression to an upgrade of dahdi/libpri, that'd be bug material - but only if you can reproduce it in 1.6 because the maintainer doesn't care about anything earlier |
| 23:03.19 | kerframil | ok, it is 1.6. fine. |
| 23:03.38 | Serees | hehe... for me it does not have to be a bug |
| 23:04.20 | Serees | i know asterisk (at least the older versions) and in my opinion me mistake is usally (at least in my case) situatied between the keyboard and the chair |
| 23:04.25 | Serees | so i assume it is the same now |
| 23:05.40 | *** join/#asterisk wonderworld (~ww@port-92-201-97-101.dynamic.qsc.de) |
| 23:06.07 | fauxalliance | checks the in-flight entertainment |
| 23:06.34 | fauxalliance | ohhh.. rofl cakes in first class |
| 23:07.42 | Serees | http://pastebin.com/KsNW9HXd |
| 23:07.47 | Serees | that's the debug output |
| 23:07.57 | Serees | don't quite understand it :s |
| 23:08.08 | fauxalliance | has to close the lid for takeoff... god bless znc |
| 23:08.51 | Serees | fauxalliance have a nice flight |
| 23:08.53 | Serees | lol |
| 23:09.39 | fauxalliance | thx1138 Serees |
| 23:12.20 | *** join/#asterisk bmg505 (~leon@196-209-44-243.dynamic.isadsl.co.za) |
| 23:13.13 | WIMPy | Serees: It doesn't even seem to try to answer the call. |
| 23:13.26 | fauxalliance | SWG456 cross check and prepare for departure ;) |
| 23:13.39 | fauxalliance | wheeeeeee! |
| 23:14.18 | Serees | what does the state 3(Establish awaiting TEI) mean? |
| 23:14.23 | Serees | it seems to be stuck there |
| 23:15.26 | WIMPy | Maybe you've got the wrong signalling. Is it really PTMP? |
| 23:15.48 | Serees | that used to work with zaptel |
| 23:17.23 | WIMPy | Well, from the called number looks likely to be ptmp, but as it can't receive a TEI, it's either PTP or the network doesn't receive what you send. |
| 23:18.35 | WIMPy | Did that setup work before? Same hardware? |
| 23:18.55 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 23:19.21 | Serees | yes, same hardware, nearly same config (only needed to 'upgrade' the dialing plan as some applications are no longer avail in 1.6) |
| 23:19.30 | WIMPy | Is that card the only device connected to that line? |
| 23:19.44 | Serees | yes |
| 23:20.23 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 23:20.45 | Serees | I changed the signaling... now it does not seems to get the D channel anymore |
| 23:21.11 | WIMPy | To bri_cpe? |
| 23:21.18 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 23:21.37 | *** join/#asterisk AMindMobile (~AMindMobi@95-28-106-82.broadband.corbina.ru) |
| 23:21.48 | Serees | no to bri_cpe_ptp ???? |
| 23:21.55 | Serees | try bri_cpe???? |
| 23:22.30 | WIMPy | There is no _ptp AFAIK. |
| 23:23.32 | Serees | hehe ok... because you where talking abbout PTP earlier i thought lest try PTP instead of PTMP :D |
| 23:23.54 | Serees | I already said it... there is a stupid link between the chair and the keyboard :p |
| 23:24.18 | WIMPy | Would make sense, but ptmp wasn't avaliable in earlier versions, so ptp is implied. |
| 23:24.27 | *** join/#asterisk cashback (~cashback@ip68-2-140-46.ph.ph.cox.net) |
| 23:26.10 | Serees | no luck |
| 23:26.27 | Serees | if i use bri_cpe it does not recognize the call even |
| 23:26.35 | Serees | so it must be bri_cpe_ptmp |
| 23:27.10 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 23:27.29 | Serees | so it is not the signalling... maybe the switchtype.... but that used to be correct as well |
| 23:28.36 | Serees | i find it very strange that the exact same setup worked with 1.2 and stopped working with 1.6... I didn't even touch the hardware :s |
| 23:28.50 | WIMPy | I'd make sure to use at least libpri 1.4.12. There have been a lot of issues with earlier versions. |
| 23:29.06 | WIMPy | Or not use libpri :-) |
| 23:29.30 | Serees | how can I see if or not i use libpri? |
| 23:29.45 | WIMPy | But that's a choice of what features you need/want. |
| 23:29.57 | WIMPy | You do. chan_dahdi needs libpri. |
| 23:30.13 | Serees | oh ok |
| 23:30.23 | WIMPy | But if you use DSS-1 you've got chan_misdn or chan_lcr as alternatives. |
| 23:30.33 | WIMPy | They don't use libpri. |
| 23:31.53 | Serees | well... I don't need a lot of features.... just to be able to make a call (if possible with setting the CID) and to answer calles (also based on the dailed number) 3 inwards dialing numbers have 3 different configs |
| 23:32.05 | Serees | other than that it does not matter to much to me |
| 23:32.44 | Serees | libpri-1.4.11.4 is installed |
| 23:32.46 | WIMPy | Well. "make a call" might not be as straight forward as one would like to think. |
| 23:33.24 | WIMPy | IIRC the 1.4.11 region had changes to TEI management as well. |
| 23:33.44 | Serees | hehe lol... i'm sure it isn't... but the wonder full developers of asterisk make it as straidforward for me as possible :D |
| 23:33.55 | Serees | lets see if I can easily upgrade that |
| 23:34.02 | WIMPy | Err, no. |
| 23:34.34 | Serees | no to easilly upgrade libpri or the other comment? |
| 23:35.00 | WIMPy | Just make a call where you've got the hole number at once and you just have a conversation is ok, but if you want to dial digit by digit or get tones and announcements it can become quite tricky. |
| 23:35.32 | Serees | i can upgrage to 1.4.12_beta2 |
| 23:35.43 | WIMPy | You need to recompile Asterisk after installing a new libpri. |
| 23:35.58 | Serees | that's not an issue |
| 23:36.45 | *** join/#asterisk RypPn2 (~RypPn2@195.149.22.58) |
| 23:37.06 | Serees | it would bee libpri 1.4.12_beta2 and asterisk-1.6.2.17 |
| 23:38.34 | *** part/#asterisk millsu2 (~brad@mail.serverplus.com) |
| 23:39.27 | Serees | libpri upgraded... asterisk recompiling |
| 23:39.53 | WIMPy | Let's hope that helps. |
| 23:41.29 | Serees | hehe, indeed |
| 23:42.26 | Serees | one of the things i was never able to get running in my previous setup was to set the outwords CID... any idea what i could have done wronge... I used the SETCID command to set it to the correct number |
| 23:42.50 | Serees | perhaps it is to difficult to say without additional data... but is that the correct way of doing it? |
| 23:43.05 | Serees | (applications are compiling) |
| 23:45.29 | WIMPy | That doesn't exist any more. Set(CALLERID(num)=...) |
| 23:45.56 | Serees | but it also works for outging calls? |
| 23:46.07 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:46.10 | WIMPy | And a possible isse is with number plans. Bot as configured in chan_dahdi.conf as well as the format in the above statement. |
| 23:46.41 | WIMPy | Where else should a set make a change? |
| 23:47.34 | Serees | Well I also use it to make a change in an incomming call.... |
| 23:47.53 | WIMPy | No, you do it on outgoing call. |
| 23:48.02 | WIMPy | You have to see it from the Asterisk perspective. |
| 23:48.06 | ruben23 | hi guys anyone have idea ekiga have g711 codec..? |
| 23:48.10 | WIMPy | It's out of Asterisk. |
| 23:48.22 | WIMPy | Everything has G.711. |
| 23:48.33 | Serees | oh ok... in that way |
| 23:48.53 | Serees | ic |
| 23:49.00 | WIMPy | Maybe only a-law or only µ-law, but certainly one of them. |
| 23:49.03 | Serees | never looked at it in that way |
| 23:49.45 | Serees | compiled and running again |
| 23:49.51 | WIMPy | Asterisk doesn't make a difference between a line or trunk and a phone. |
| 23:49.52 | Serees | let's have a look |
| 23:50.16 | Serees | it acts as a bridge |
| 23:50.22 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 23:50.34 | Serees | call is not detected |
| 23:50.40 | Serees | strange |
| 23:53.55 | ruben23 | WIMPy: but on the ekiga softphones codec menu there is no codec present g711. |
| 23:54.06 | *** join/#asterisk RypPn2 (~RypPn2@rosscom.co.uk) |
| 23:55.12 | WIMPy | ruben23: It's called PCMA or ACMU there. |
| 23:56.38 | ruben23 | WIMPy: thanks, i have registered the ekiga on my asterisk already- how do i make conferecen call on this any idea..? |
| 23:57.31 | WIMPy | Use MeetMe() or ConfBridge(). |
| 23:58.18 | Serees | it does not load chan_dahdi.so ... configure it in modules.conf? autoload=yes |
| 23:58.48 | WIMPy | Maybe there's something wrong in chan_dahdi.conf? |
| 23:58.51 | ruben23 | <PROTECTED> |
| 23:59.15 | WIMPy | If you turn up debug/verbose and try to manually load it you should get a message. |
| 23:59.32 | Serees | no.... asterisk.conf is missing with the directory to load modules from.... reinstalling it messed up my config it seems |
| 23:59.39 | WIMPy | ruben23: You put that in Asterisk's dialplan. |