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| 03:32.50 | SirLagz | is there a way to force asterisk to hang a call up? |
| 03:36.01 | SirLagz | wait, figured out the right terms to google :) |
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| 05:45.40 | ruben23 | hi guys any idea with this error log flooding on my asterisk ---> http://pastebin.com/qdqqQqWX |
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| 07:09.59 | iulhk | i have a sip account A & B, B is offline, I hold A call unless B came online , Now A call is in loop unless time reach or user sip B came online, during this loop if sip A disconnect my AGI doesn't hangup, how can i get current channel status if user A hangs the call? |
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| 07:25.17 | dsr | Hi, I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the user and call AgentLogin. But after that when I call AgentRequest I keep getting Agent '1234' is busy. If I put a delay of 5 second or more before calling AgentRequest then it works most of the times. Any idea what maybe wrong? |
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| 07:38.01 | iulhk | can i use sip show inuse command in agi ? |
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| 08:55.23 | dsr | Hi, I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the user and call AgentLogin. But after that when I call AgentRequest I keep getting Agent '1234' is busy. If I put a delay of 5 second or more before calling AgentRequest then it works most of the times. Any idea what maybe wrong? |
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| 09:05.36 | iulhk | can i use sip show inuse command in agi ? |
| 09:13.24 | BlackDex | Hello there... |
| 09:13.59 | BlackDex | i was debugging something yesterday, and a saw a lots of "501 Method Not Implement" messages and they all come from 3 phones |
| 09:14.41 | BlackDex | all other phones seem to not have this problem |
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| 09:26.26 | BlackDex | it seems to have something to do with DND option |
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| 11:28.09 | hexanol | by the way the channel topic doesn't seem up to date; shouldn't it say "DAHDI 2.9.2" instead of 2.9.0 ? :) |
| 11:31.26 | Chainsaw | There is a 2.9.2?! |
| 11:31.46 | hexanol | Chainsaw: look for yourself http://asterisk.org/downloads/dahdi |
| 11:32.03 | Chainsaw | hexanol: I rely on the channel topic in #asterisk |
| 11:32.35 | hexanol | mah you deny the reality outside of #asterisk |
| 11:32.48 | Chainsaw | hexanol: The website has no convenient location that has all version numbers I care about in one place. |
| 11:33.09 | Chainsaw | hexanol: If it did, I would check that. |
| 11:33.25 | hexanol | there's even an asterisk 13.0.0-beta1 version right now |
| 11:34.18 | hexanol | well I guess you could follow the asterisk-announce mailing list |
| 11:34.24 | hexanol | http://lists.digium.com/pipermail/asterisk-announce/ |
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| 12:54.58 | Stefan27 | how can i probe the state of a SIP device from CLI? as in "show device_state(SIP/mydevice300)" |
| 12:55.03 | Stefan27 | devstate list only works for custom devices |
| 12:58.10 | [TK]D-Fender | devstate works for SIP devices as well |
| 12:58.26 | [TK]D-Fender | Actually.. not sure on CLI |
| 12:58.31 | [TK]D-Fender | was thinking in the dialplan |
| 12:58.42 | [TK]D-Fender | make a hint for it |
| 12:58.56 | [TK]D-Fender | and you can check it that way... or just look if there are channels listed for it |
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| 13:30.05 | ccha2 | hello, is it possible to use a modem card instead of a FXO card ? |
| 13:31.13 | lorenzo | Hi! I've set up my first phone number on asterisk, but I'm getting odd logins since I enabled allowguest=yes |
| 13:31.40 | DruidZ | Logins or login attempts? |
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| 13:31.47 | lorenzo | <PROTECTED> |
| 13:31.50 | lorenzo | <PROTECTED> |
| 13:32.07 | lorenzo | what does that mean? |
| 13:32.08 | WIMPy | ccha2: No |
| 13:32.13 | lorenzo | I see it's a nepal number in blacklist somewhere |
| 13:32.34 | WIMPy | lorenzo: People are trying to make calls through you. |
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| 13:33.05 | DruidZ | I am asking if they are actually registering or just attempting and failing. |
| 13:33.13 | lorenzo | so they're basically phoning me and then placing an extension as number? |
| 13:33.20 | lorenzo | (a number as extension) |
| 13:33.23 | lorenzo | in order to reach out |
| 13:33.34 | WIMPy | yes |
| 13:33.36 | ccha2 | is it possible to make voice calls with a modem card ? |
| 13:33.50 | DruidZ | The real question is, did you remove all the example numbers or at least change all the passwords. |
| 13:33.56 | lorenzo | DruidZ: I see no SIP login attempts |
| 13:34.08 | WIMPy | ccha2: Usuall yes, but not with Asterisk. |
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| 13:36.03 | iulhk | can i use sip show inuse command in agi ? |
| 13:36.21 | lorenzo | DruidZ: no :/ you mean the iaxtel700, trunkint stuff? |
| 13:36.48 | WIMPy | iulhk: You can use asterisk -x. |
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| 13:39.19 | lorenzo | I see there are tons of default extensions in dialplan show |
| 13:39.56 | WIMPy | Don't start Asterisk with the sample configs if you're connected to the internet. |
| 13:40.52 | DruidZ | lorenzo: Let's step back a moment. When you say "set up my first phone number" exactly what do you mean? Are you talking about an entry in sip.conf to register your phone, an entry in extensions.conf so that people can call you or both? |
| 13:41.09 | lorenzo | DruidZ: I've set it inbound only as I didn't trust this setup |
| 13:41.13 | lorenzo | so I have a register in sip.conf |
| 13:41.30 | lorenzo | I can phone in and that's it for now, but I'd like to phone out too, hence my need to secure this thing |
| 13:41.38 | iulhk | <WIMPy>: ok, so can i use "sip show inuse" for some specific peer ? and get specific data either peer is busy or not? don't want all list ? |
| 13:41.49 | DruidZ | What WIMPy said. |
| 13:43.15 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-qnxqvfurrmixufht) |
| 13:43.24 | DruidZ | What does the last line of "sip show peers" say? |
| 13:43.44 | WIMPy | iulhk: If you only want wan you can use the DEVICE_STATE function with get full variable. |
| 13:43.56 | Katty | infobot: crittercam |
| 13:43.56 | infobot | methinks crittercam is Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
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| 13:56.41 | msaraiva | Hi all |
| 13:56.56 | msaraiva | I'm having a bit of a problem with chan_pjsip |
| 13:57.01 | msaraiva | Can't load the module |
| 13:57.06 | msaraiva | WARNING[16002]: loader.c:565 load_dynamic_module: Error loading module 'chan_pjsip.so': /usr/lib/asterisk/modules/chan_pjsip.so: undefined symbol: ast_sip_send_request |
| 13:57.13 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 13:57.20 | msaraiva | I got pjproject from Asterisk's git repo |
| 13:57.43 | file | the PJSIP stuff is made up of many modules, not just chan_pjsip.so |
| 13:57.51 | file | res_pjsip.so and tons more |
| 13:58.09 | Stefan27 | sip show peer erikmaindesk62 says "... Callerid : "" <24> ... " how come the generated Status events (AMI) have callerid unknown? the event says: "Event: Status, Privilege: Call, ...,Channel: SIP/erikmaindesk62-0000005234, CallerIDNum:<unknown>,CallerIDNAME: <unknown>, ..., |
| 13:58.11 | msaraiva | Humm |
| 13:58.21 | msaraiva | Just tried loading res_pjsip and Asterisk crashed |
| 13:58.43 | Stefan27 | Moreover, the Status event also has ConnectedLineNum: <unknown> |
| 13:59.29 | Stefan27 | any ideas what could make fields in manager events <unknown>? |
| 14:00.27 | WIMPy | What event exactely? |
| 14:00.45 | msaraiva | I guess the problem is that i'm re-using my modules.conf from Asterisk 11, so not all modules are being loaded for PJSIP to work |
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| 14:03.51 | Stefan27 | let me try paste it |
| 14:04.24 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-hsdrlnthdlngrfkj) |
| 14:04.39 | Stefan27 | but it's an event of type Status |
| 14:05.54 | Stefan27 | http://pastebin.com/vbVC8wPA |
| 14:08.55 | msaraiva | Ok, some progress, i guess |
| 14:09.07 | msaraiva | Now i have an error while loading res_pjsip_pubsub.so |
| 14:09.09 | msaraiva | asterisk: ../src/pjsip-simple/evsub.c:296: pjsip_evsub_init_module: Assertion `endpt != ((void *)0)' failed. |
| 14:09.44 | ThatsKP | generally question... in the v11 docs it say Asterisk GUI should not be used. Is there another GUI that can be used at this time? Also, what would be the risk of continuing to use the GUI for now? |
| 14:09.48 | WIMPy | Stefan27: Is that *12? |
| 14:10.51 | WIMPy | ThatsKP: There are many GUIs. The only one known here to have some support is FreePBX. |
| 14:10.59 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-uglogpperonfdohj) |
| 14:11.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:11.06 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 14:12.12 | ThatsKP | I thought FreePBX was a complete offering. Its possible to just use their GUI? |
| 14:12.26 | jameswf | is a fan of FreePBX |
| 14:12.50 | jameswf | ThatsKP: both distro and independent |
| 14:12.58 | WIMPy | They do have a complete distro now as well, but it is "only" a GUI. |
| 14:13.06 | ThatsKP | no kidding |
| 14:13.18 | ThatsKP | their website came across was the exactly opposite of that |
| 14:13.27 | WIMPy | For more information join #freepbx. |
| 14:14.31 | file | msaraiva, I'd suggest doing autoload as a test to make sure your modules are sound |
| 14:14.32 | ThatsKP | cool they have a channel- I'll dl it can give it go first |
| 14:16.48 | Stefan27 | no the event is generated due to an internal call from a sip device to another sip device (i think) |
| 14:20.07 | msaraiva | @file, thanks. Just tried that, it works with all conf files from samples and autoload |
| 14:20.54 | file | msaraiva, well that's good - now you just need to construct the list in order |
| 14:20.59 | msaraiva | Now i need to figure out which conf files and modules are needed |
| 14:21.41 | msaraiva | Yeah...i'm really fond of leaving autoload on and having a bunch of config files that are not needed |
| 14:22.00 | msaraiva | It's a different beast from Asterisk 11 |
| 14:23.25 | msaraiva | On a related note, i'm trying to 12 to see if PJSIP behaves differently than chan_sip on one specific instance - re-invite mid call to change media ip |
| 14:24.16 | msaraiva | On Asterisk 11 / 1.8, this specific setup results in one-way audio (Asterisk is proxying media, no NAT involved) |
| 14:24.47 | msaraiva | Took many traces but couldn't find anything wrong |
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| 14:26.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:33.49 | msaraiva | @file, managed to streamline modules.conf with PJSIP. Seems the order doesn't really matter, as long as all needed modules are there. FYI, i was missing a couple of resource modules for sorcery. |
| 14:33.51 | msaraiva | Thanks again. |
| 14:34.18 | file | load all the things! |
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| 14:36.13 | msaraiva | My OCD won't let me. :S |
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| 14:52.09 | geoid | greetz! |
| 14:52.38 | geoid | I've got a file /etc/asterisk/sip_provisioning.conf |
| 14:52.57 | geoid | This file sets the defaults for when phones connect to asterisk for provisioning. |
| 14:53.12 | geoid | using the [Provisioning] config group |
| 14:53.22 | geoid | however, I can't find any reference to this in the documentation |
| 14:53.38 | geoid | I'm trying to set the default language files for provisioning |
| 14:53.52 | geoid | Any clues? |
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| 15:06.03 | Stefan27 | found the problem, callerid was destroyed by a agi-script before the dial |
| 15:08.39 | infernix | i have asterisk video working, but when a non-video sip client calls a video sip client, it still tries to negotiate video |
| 15:09.06 | infernix | is there some way to disable video to outgoing Dial() commands in the dialplan? |
| 15:09.22 | infernix | maybe disallow certain codecs, but it has to be conditionally - not sure if that can be done in the dialplan at all? |
| 15:11.19 | WIMPy | Create two peers for the device with different codec lists. |
| 15:11.47 | infernix | so phone1-video and phone1-audio? |
| 15:12.09 | *** part/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
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| 15:12.24 | WIMPy | Thats the best idea I have. |
| 15:12.29 | infernix | but then the device has to register twice, right? |
| 15:12.41 | hexanol | mjordan: just to let you know, the topic for this channel is not up to date concerning DAHDI |
| 15:12.58 | WIMPy | If it has to register, yes. |
| 15:13.34 | infernix | happens to be impossible for the webclient in use (sipml5) |
| 15:21.49 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-toqnchvsrsrfywvc) |
| 15:26.14 | infernix | so the contact header seems to define 'video' as per rfc 3840 |
| 15:26.30 | infernix | is there some way to alter the contact header before a dial is being sent? |
| 15:30.39 | *** join/#asterisk theron_ (~theron@69.63.185.56) |
| 15:34.34 | *** join/#asterisk theron (~theron@184.sub-70-197-193.myvzw.com) |
| 15:57.58 | msaraiva | bridge.c:772 bridge_base_init: Bridge 4457ae56-e046-40d5-8ef6-982df7647844: Could not create class basic. No technology to support it. |
| 15:58.08 | msaraiva | Anyone got that on Asterisk 12? |
| 15:58.27 | msaraiva | Using PJSIP |
| 16:03.37 | msaraiva | Nevermind, just being dumb. |
| 16:05.42 | *** join/#asterisk keiths_ (~keiths@goatfarm.vianet.ca) |
| 16:10.41 | mjordan | hexanol: thanks. My DAHDI involvement is rather limited, so it probably doesn't get updated until someone notices |
| 16:12.27 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.9.2 (2014/07/21), DAHDI-tools 2.9.2 (2014/07/21); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 16:14.47 | msaraiva | Well, PJSIP has worked it's first miracle |
| 16:14.55 | msaraiva | My one-way audio problem is gone. |
| 16:15.09 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 16:15.09 | msaraiva | Thank you, Mr. Asterisk Gurus. |
| 16:21.54 | Katty | today is jblack's bday! |
| 16:21.58 | Katty | infobot: seen jblack |
| 16:22.05 | infobot | jblack <~jblack@73.181.147.102> was last seen on IRC in channel #asterisk, 92d 10h 6m 59s ago, saying: 'bhavikpatel6842: THey want you to paste your logs to pastebin.com instead of here.'. |
| 16:23.50 | mjordan | msaraiva: yay? (most likely either putnopvut or file, honestly) |
| 16:24.08 | file | nothing to see here, move along |
| 16:28.08 | *** join/#asterisk voip (~poppa@asterisk.phonesupport.org) |
| 16:28.53 | lorenzo | okay, I've removed all the default extensions |
| 16:28.59 | lorenzo | I still get people doing this kind of dirty stuff |
| 16:29.11 | lorenzo | <PROTECTED> |
| 16:29.24 | *** join/#asterisk v0ip (~poppa@asterisk.phonesupport.org) |
| 16:29.26 | lorenzo | I guess it's just like the real world, people tring to steal etc :/ |
| 16:29.31 | file | yes. |
| 16:29.52 | *** join/#asterisk kaziklu-bey (0c0adbde@gateway/web/freenode/ip.12.10.219.222) |
| 16:30.04 | lorenzo | I'm getting two kinds of messages, one is that, which I know is a reject |
| 16:30.17 | lorenzo | the other is this one |
| 16:30.17 | lorenzo | <PROTECTED> |
| 16:30.52 | lorenzo | I've read here it means that they're trying to dial through my machine but how? |
| 16:31.54 | file | they send a call to machine and due to your configuration it goes to a context where stuff happens? |
| 16:32.39 | file | since July 15th (of this year) my public facing Asterisk 12 (soon to be 13) has had 8188869 failed authentication attempts |
| 16:32.46 | lorenzo | I have a trap for anything that is not in context |
| 16:32.53 | lorenzo | maybe this is what is being answered |
| 16:32.57 | *** join/#asterisk Tagor (53567e1f@gateway/web/freenode/ip.83.86.126.31) |
| 16:34.53 | lorenzo | <PROTECTED> |
| 16:34.56 | lorenzo | oh yeah |
| 16:35.06 | lorenzo | :D |
| 16:35.28 | Tagor | does anyone know if you can run ASR() and DTMF together? i would like to make an option 'say or enter the extensions number' |
| 16:35.50 | [TK]D-Fender | [12:30]lorenzoI've read here it means that they're trying to dial through my machine but how? <- you're allowing un-authed calls. Stop. |
| 16:36.22 | lorenzo | hm, is there a way to disable that and still allow to be reachable from a voip number? |
| 16:36.25 | lorenzo | a did number |
| 16:36.41 | [TK]D-Fender | [general] allowguest=no <-------- |
| 16:37.19 | lorenzo | yeah, but then it rejects calls from landine/mobile users who try to reach the pbx |
| 16:37.27 | lorenzo | there must be some sort of whitelist |
| 16:37.34 | [TK]D-Fender | only if you don't set your peer for your provider properly |
| 16:44.10 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 16:47.46 | lorenzo | hmm |
| 16:47.52 | lorenzo | looks like my provider is using insercure=very |
| 16:48.01 | lorenzo | which asterisk 11 rejects |
| 16:49.32 | [TK]D-Fender | pastebin your peer and the call masking only the secret |
| 16:49.37 | [TK]D-Fender | ~pb |
| 16:49.37 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:49.39 | [TK]D-Fender | ^^^ |
| 16:49.48 | [TK]D-Fender | with "sip set debug on" <- |
| 16:50.38 | lorenzo | http://pastebin.com/iWLjZm6q |
| 16:51.47 | *** join/#asterisk gugaua (~gugaua@unaffiliated/gugaua) |
| 16:52.09 | gugaua | Hello, I want to use Asterisknow with a Frotz ISDN card, is that possible? |
| 16:52.45 | [TK]D-Fender | lorenzo: insecure=port,invite , nat=no, directmedia=no |
| 16:53.11 | [TK]D-Fender | lorenzo: And it's rare to need to specify a port like that... test without it afterwards |
| 16:53.54 | lorenzo | insecure=port,invite alone did the trick :)) |
| 16:53.57 | [TK]D-Fender | gugaua: Asterisk support AVM Fritz cards |
| 16:54.05 | [TK]D-Fender | lorenzo: Do the other settings as well. |
| 16:54.41 | gugaua | [TK]D-Fender: okay thats great :) I will install asterisknow now! ;) |
| 16:55.00 | lorenzo | I already have canreinvite=no (which is now directmedia) and nat=no in my general |
| 16:55.07 | lorenzo | is it redundant to add it again? |
| 16:56.23 | lorenzo | more generally, do other users inherit the general if there's no override? |
| 16:56.31 | [TK]D-Fender | Exactly |
| 16:56.38 | lorenzo | perfect |
| 16:56.40 | [TK]D-Fender | Always set your peers explicitly |
| 16:56.55 | [TK]D-Fender | Regardless of if they happen to match. |
| 16:57.40 | lorenzo | ok, thanks |
| 16:57.46 | lorenzo | another dumb question incoming :P |
| 16:57.50 | lorenzo | if I have SIP access to another PBX |
| 16:58.00 | lorenzo | can I register it in asterisk, and dial through asterisk? |
| 16:58.13 | [TK]D-Fender | that is no different that the provider you already have |
| 16:58.17 | [TK]D-Fender | SIP is SIP is SIP. |
| 16:58.47 | lorenzo | okay, what's the difference between having register => and e.g. the [sip.messagenet.it] |
| 16:58.50 | lorenzo | section |
| 16:59.20 | lorenzo | I see that the section between brackets is used for bouncing users to the appropriate context, when calling me from that host |
| 16:59.30 | lorenzo | but isn't register => enough if I want to make outbound calls only? |
| 16:59.34 | [TK]D-Fender | lorenzo: One is an entry that matches calls from a source, and defines how you set up auth and where to send calls. |
| 16:59.37 | [TK]D-Fender | ~sipregister |
| 16:59.37 | infobot | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 16:59.39 | [TK]D-Fender | ^^^^^ |
| 17:02.13 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-hsdomaugjozkyukq) |
| 17:02.56 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:04.01 | lorenzo | okay thanks |
| 17:04.11 | lorenzo | I also see that calls incoming from sip registratons |
| 17:04.20 | lorenzo | are placed into the context specified in the section with brackets |
| 17:04.22 | lorenzo | is that right? |
| 17:04.37 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 17:04.56 | [TK]D-Fender | calls don't come from registrations |
| 17:05.00 | [TK]D-Fender | call comes from sources |
| 17:05.15 | [TK]D-Fender | they may know where to reach you beacause YOU registered to them... that's another matter |
| 17:05.28 | lorenzo | I see |
| 17:05.36 | [TK]D-Fender | Calls are matched against your devices definitions inyour configs. |
| 17:09.37 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 17:10.24 | lorenzo | yeah it's all working now thanks ^_^ |
| 17:10.24 | *** join/#asterisk jhe5ter (~jhester@nat/digium/x-rtpthasticfpeknq) |
| 17:10.29 | lorenzo | I've set up three trunks |
| 17:10.40 | lorenzo | I hope I don't end up with a million dollar bill o_o |
| 17:11.40 | jeffspeff | i want to be able to receive sms to any of my DID's and optionally send SMS as well. I've googled some sms gateways, but they seem to have/charge for a lot more features and functions that i either don't need or would accomplish in my dialplan. Is that just what I'm stuck with in the U.S. or is that something that you usually get through your SIP provider (Intelepeer) ? |
| 17:13.29 | file | jeffspeff, you may or may not get it with your SIP provider - and how they do it is up to them |
| 17:14.30 | gusto | WIMPy, hi |
| 17:14.34 | gusto | WIMPy, how are you? |
| 17:14.40 | gusto | Penguin, hi |
| 17:14.43 | gusto | Penguin, how are you? |
| 17:15.50 | *** join/#asterisk absentbird (~shawn@216-235-104-HCC-187.hcc.net) |
| 17:15.53 | jeffspeff | file, how is this usually accomplished? |
| 17:16.24 | [TK]D-Fender | jeffspeff: Depends on your provider |
| 17:16.43 | [TK]D-Fender | jeffspeff: If you send an SMS to a number they dilver to you, do you see anything in the SIP DEBUG? |
| 17:16.52 | [TK]D-Fender | jeffspeff: If not.. then that's the end of it. |
| 17:17.00 | absentbird | I restarted my asterisk server and now every call says "contact your administrator for permissions." Does anyone know why this is happening? |
| 17:17.04 | jeffspeff | I haven't tried yet, just gathering information on it. |
| 17:17.32 | jeffspeff | [TK]D-Fender, SMS is dependant on my provider regardless of me using them as an SMS gateway? |
| 17:19.02 | file | I would say these days the providers that do SMS are the minority |
| 17:19.29 | [TK]D-Fender | jeffspeff: How are you using them as an SMS gateway ... if you aren't using them as an SMS gateway and are asking us? |
| 17:21.07 | jeffspeff | [TK]D-Fender, i'm trying to learn how it's typically done. am i restricted to only going through my sip provider? are there some recommended sms gateway companies out there that don't charge for a lot of extra unused features? |
| 17:22.14 | [TK]D-Fender | jeffspeff: I'm not sure if you can use other providers to spoof a number they don't really host for you... and that is for sending alone... |
| 17:22.25 | [TK]D-Fender | Recieving would clearly land at your current provider... |
| 17:22.37 | [TK]D-Fender | jeffspeff: `So go try, go ask them.. and then reconsider your options... |
| 17:22.37 | absentbird | In the logs I have this error: ast_carefulwrite: write() returned error: Connection refused |
| 17:23.41 | *** part/#asterisk karlh626 (~karlh626@addr-199.21.194.73.nptpop-cmts-cable-sub.rdns-bnin.net) |
| 17:24.02 | [TK]D-Fender | absentbird: Sounds like a pporly written AGI writing out of order. |
| 17:24.39 | absentbird | [TK]D-Fender, how should I proceed? |
| 17:25.09 | [TK]D-Fender | absentbird: Confirm where you actually get it |
| 17:25.23 | *** join/#asterisk jhester (~jhester@mobile-166-147-115-093.mycingular.net) |
| 17:25.35 | absentbird | Every time I dial a number. |
| 17:25.50 | absentbird | I mean, every time I try to make a call |
| 17:25.55 | *** join/#asterisk jhester (~jhester@nat/digium/x-jctkdnegzqtdrztu) |
| 17:26.23 | absentbird | a voice says "Contact your administrator for permissions" and that error is in the logs |
| 17:26.52 | *** join/#asterisk jhe5ter (~jhester@mobile-166-147-115-093.mycingular.net) |
| 17:27.09 | absentbird | I also get this warning: launch_netscript: Connect to 'agi://127.0.0.1:6969/tollrest?type=extension&content=105&server=SIP/00085D2459FF' failed: Connection refused |
| 17:28.59 | [TK]D-Fender | sounds like you have a socket-based AGI listener on your system... go figure out what you have and how it's set up and supposed to be working. |
| 17:29.18 | [TK]D-Fender | You and prove that it's listening as well... |
| 17:30.17 | absentbird | I am pretty new to this, what would be the first step for investigating this? |
| 17:31.36 | [TK]D-Fender | I just told you |
| 17:31.51 | [TK]D-Fender | Go find out what process is supposed to be listening on that port |
| 17:31.57 | *** join/#asterisk zamba (marius@flage.org) |
| 17:31.57 | absentbird | ok |
| 17:32.01 | [TK]D-Fender | We have no idea what you're running. |
| 17:34.58 | absentbird | Well nothing is listening on port 6969 |
| 17:35.06 | [TK]D-Fender | Go find out what should be... |
| 17:35.13 | [TK]D-Fender | check your startup scripts, etc... |
| 17:36.23 | absentbird | like /etc/init.d/asterisk? |
| 17:36.35 | [TK]D-Fender | No.. that starts Asterisk itself... |
| 17:36.42 | [TK]D-Fender | It probably shouldn't be in there... |
| 17:36.47 | [TK]D-Fender | but can't hurt I guess |
| 17:38.41 | *** join/#asterisk areski (~areski@80.174.128.86.dyn.user.ono.com) |
| 17:39.35 | *** join/#asterisk absentbird_ (~shawn@216-235-104-HCC-187.hcc.net) |
| 17:40.47 | absentbird_ | [TK]D-Fender, So where are the startup scripts? Sorry for being such a newbie. |
| 17:41.47 | *** join/#asterisk jhester (~jhester@nat/digium/x-tktpcjcqntqbgziw) |
| 17:41.52 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 17:42.44 | Tagor | does anyone know if you can run ASR() and DTMF together? i would like to make an option 'say or enter the extensions number'' |
| 17:42.53 | *** join/#asterisk jhester (~jhester@nat/digium/x-dzqwmwupsaqznwcn) |
| 17:44.06 | *** join/#asterisk jhe5ter (~jhester@nat/digium/x-pflhgmtkeecuujjf) |
| 17:45.24 | gugaua | [TK]D-Fender, I installed asterisknow and all I get is all circuits are busy now with the AVM Fritz card |
| 17:46.09 | gugaua | [TK]D-Fender the card is recognized 02:08.0 Network controller: AVM GmbH Fritz!PCI v2.0 ISDN (rev 01) |
| 17:46.29 | [TK]D-Fender | gugaua: Keep in mind... |
| 17:46.30 | [TK]D-Fender | ~freepbx |
| 17:46.31 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:46.32 | [TK]D-Fender | ^^^ |
| 17:47.01 | gugaua | ahh okay so that has "nothing" to do with asterisk? |
| 17:47.52 | [TK]D-Fender | gugaua: that message can mean a LOT of things. |
| 17:48.07 | [TK]D-Fender | gugaua: that is not inherently an asterisk issue |
| 17:48.13 | gugaua | How can I check if the card works, like d-line check |
| 17:48.25 | [TK]D-Fender | gugaua: And you don't have direct control over your system.... |
| 17:48.27 | [TK]D-Fender | USE it |
| 17:48.31 | [TK]D-Fender | testing means doing |
| 17:48.49 | [TK]D-Fender | Also depends how you set it up |
| 17:48.53 | [TK]D-Fender | and IF you set it up |
| 17:49.11 | [TK]D-Fender | If you expected it to be magically detected and configured.... you might be in for a surpriwse |
| 17:49.33 | gugaua | It´s the question where it should be detected... |
| 17:49.38 | gugaua | dahdi won´t find it |
| 17:50.06 | *** join/#asterisk Yanxi (~procube@port-92-204-112-167.dynamic.qsc.de) |
| 17:50.51 | [TK]D-Fender | gugaua: Most of what I see referes to using chan_capi which freePBX doesn't natively configure for you |
| 17:51.03 | [TK]D-Fender | WIMPy: You know some more about this series? |
| 17:54.16 | *** join/#asterisk absentbird (~shawn@216.235.104.187) |
| 17:55.00 | *** part/#asterisk Yanxi (~procube@port-92-204-112-167.dynamic.qsc.de) |
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| 18:21.28 | *** part/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 18:21.45 | lorenzo | how do you specify "do nothing" in a GoToIf? |
| 18:22.21 | kaziklu-bey | noop |
| 18:24.13 | [TK]D-Fender | literally put nothing |
| 18:29.11 | [TK]D-Fender | [14:22]kaziklu-beynoop <- will not work. |
| 18:30.24 | *** join/#asterisk koffel (~koffel@c-71-230-13-15.hsd1.nj.comcast.net) |
| 18:30.47 | koffel | hello |
| 18:30.52 | kaziklu-bey | ow |
| 18:31.06 | koffel | i got a wierd problem with 11.11 |
| 18:31.36 | [TK]D-Fender | koffel: Try converting it back from binary... |
| 18:31.41 | [TK]D-Fender | koffel: That usually helps |
| 18:31.44 | koffel | it says it mysql engine is not available |
| 18:32.03 | koffel | WARNING[3137]: config.c:2578 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 18:35.08 | [TK]D-Fender | Doesn't sound "weird" yet. Just sounds mis-configured or not all up and running... |
| 18:35.36 | koffel | dadhi and libpril and asterisk are installed |
| 18:35.39 | koffel | and running |
| 18:37.11 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 18:37.17 | [TK]D-Fender | koffel: And you've made no mention of what is using MySQL at all. DADHI sure doesn't |
| 18:38.17 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 18:39.49 | koffel | i have that setup |
| 18:42.33 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:43.11 | koffel | i gota be missing something on ubuntu 12.04 |
| 18:43.29 | koffel | i tried to load mysql module in asterisk cli |
| 18:43.33 | koffel | <PROTECTED> |
| 18:44.51 | [TK]D-Fender | I'm pretty sure no file in upper-case like that exists there... |
| 18:47.17 | *** join/#asterisk eeezkil (~eeezkil@unaffiliated/eeezkil) |
| 18:53.13 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-85-193.user.veloxzone.com.br) |
| 18:53.45 | anonymouz666 | hi all. anyone know how holdtime in app_queue is calculated? |
| 18:53.53 | anonymouz666 | which interval is used? |
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| 19:45.54 | gugaua | can someone help me to setup Fritz PCI ISDN card with asterisk? |
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| 19:56.30 | madsara | Hey, I'm upgrading to ooh323, and I'm having issues determining how to register prefixes. I used to (with the former h323.conf) be able to register ala prefix=0,1,2,3,4 ... etc... how do I register such that _everything_ should hit my asterisk install? |
| 20:11.44 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:26.00 | madsara | Ok, looks like I can just configure the prefixes in gnugk.ini instead, this seems to work |
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