| 00:00.33 | Alex25 | any trick to do that? |
| 00:02.01 | [TK]D-Fender | nothing I can think of. |
| 00:02.47 | [TK]D-Fender | other than perhaps not doing a read but instead just redirecting the call PAST the read |
| 00:03.59 | Alex25 | I see |
| 00:04.07 | Alex25 | that's a problem |
| 00:06.57 | Alex25 | is it possible to execute a GOTO using AMI? |
| 00:07.45 | Alex25 | or using some script? |
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| 00:10.06 | mjordan | A redirect is essentially a GoTo. |
| 00:10.37 | TazzNZ | telnet is just a socket connection - how can it be evil ? |
| 00:10.37 | mjordan | If what you want to do is have a channel automagically bypass some entering of DTMF, you're best of redirecting it to where you want, rather than attempting to emulate the user pressing DTMF |
| 00:10.48 | mjordan | TazzNZ: Have you read the RFC? :-) |
| 00:11.13 | file | triggers another, potentially |
| 00:11.13 | Alex25 | how can i force redirection to another priority using an external script? |
| 00:11.14 | file | Texas |
| 00:11.14 | Nugget | Don't mess with Texas. |
| 00:11.38 | mjordan | Alex25: AMI Redirect. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Redirect |
| 00:12.22 | Alex25 | thank you |
| 00:12.43 | TazzNZ | mjordan: I'd have to say no. Other than not having encryption, I can't see the evilness :) |
| 00:14.10 | Alex25 | btw someone has apparently managed to solve my problem by editing source code http://forums.asterisk.org/viewtopic.php?f=1&t=84160&start=0 |
| 00:14.26 | Alex25 | unfortunately he hasn't posted his work |
| 00:14.51 | Alex25 | TazzNZ: I'm using telnet only internally |
| 00:15.42 | mjordan | Let's just say there is more to it than a socket :-) |
| 00:17.00 | mjordan | Alex25: As David pointed out on that issue, if what you want to do is programatically inject DTMF into Asterisk, a Local channel works just fine. We use them all the time to exercise Asterisk in the Test Suite. But pretending like you're a SIP device and are passing DTMF? Nope. Probably not going to happen. |
| 00:17.55 | TazzNZ | mjordan: agreed - but most people use it in a very limited way "Is that server listening?" and/or switches - the good old glory days of telnet died :( |
| 00:20.03 | Alex25 | mjordan: I know but there must be another way to do that.. |
| 00:23.50 | *** join/#asterisk undecided (~kvirc@122.162.104.171) |
| 00:24.46 | mjordan | nope. |
| 00:25.00 | mjordan | You can either use a Local channel, or you can redirect the channel out of Read and put them where you want |
| 00:26.12 | [TK]D-Fender | actually.. maybe a local bridged in MIGHT work. |
| 00:26.39 | mjordan | oh, you can get ugly :-) |
| 00:26.51 | mjordan | SIP <-- ConfBridge -> Local;1 <-> Local;2 <-- Read |
| 00:27.01 | mjordan | You can redirect the Local;2 to an extension that does SendDTMF |
| 00:27.08 | mjordan | then redirect it back into the ConfBridge |
| 00:27.10 | mjordan | voila. |
| 00:27.17 | mjordan | But that is ... ew. |
| 00:27.57 | undecided | thank you for ideas. I'm trying the redirect first |
| 00:28.26 | undecided | hope that won't mess my variables |
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| 00:39.22 | [TK]D-Fender | mjordan: Oh god... split the channel? |
| 00:39.40 | [TK]D-Fender | mjordan: dual redirect? That would be dirty as all get out... |
| 00:40.29 | [TK]D-Fender | Checkout time (was HOURS ago.. but trapped with nasty switch issues) |
| 00:40.34 | [TK]D-Fender | heads out |
| 00:52.11 | undecided | OK |
| 00:52.46 | undecided | AMI redirect worked for me |
| 00:53.12 | undecided | many thanks to you :) |
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| 02:01.21 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 02:05.40 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 02:05.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 04:33.30 | snadge | this is going to sound pretty dumb.. but what does the hold button do on a voip phone |
| 04:33.40 | snadge | with regards to how that interacts with asterisk |
| 04:33.44 | ChannelZ | Puts someone on hold. |
| 04:34.05 | snadge | yeah i know that.. i guess im interested in the mechanism of how that works |
| 04:34.10 | ChannelZ | I redirects them to a stream of music. |
| 04:34.19 | ChannelZ | Watch a SIP debug, you'll see |
| 04:34.28 | snadge | yeah i have some pbx software, which can select different hold music depending on the inbound route |
| 04:34.42 | snadge | but thats only whilst they're in the queue |
| 04:34.56 | snadge | if they manually press the hold button.. there can only be one default music stream |
| 04:35.24 | ChannelZ | there are MOH contexts just like anything else |
| 04:36.00 | snadge | but i want it to select the same hold music as the inbound route.. presumably thats only going to work on inbound calls though.. how it would know for an outbound, i have no idea |
| 04:36.16 | snadge | unless you used a prefix to dial out |
| 04:36.29 | snadge | which is what some people do for setting callerid etc |
| 04:36.37 | ChannelZ | You can set the MOH class in your dialplan dynamically if you want |
| 04:36.52 | ChannelZ | CHANNEL(musicclass) |
| 04:40.17 | snadge | yeah im just looking for MOH class.. its a thirdlane pbx.. i know.. get out.. there's the door etc.. sigh :P |
| 04:40.49 | ChannelZ | You can stay, I just can't help you much. |
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| 05:30.50 | snadge | yeah thats alright.. you've given me enough to make something up to tell the customer anyway and put it off until later |
| 05:31.25 | snadge | the usual technical boffins when it comes to these sorts of hairy questions.. are both not here at the moment.. great planning ;) |
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| 07:12.02 | dym | Hey. I've connected a Speedphone (AVM) to my asterisk and now on incoming calls i have a weird very loud noise. (it's not the woman on the other end shouting) ulaw, alaw and gsm are enabled. Any idea? |
| 07:12.08 | dym | Outgoing calls are mighty fine |
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| 09:59.00 | andycol | hi guys |
| 09:59.06 | andycol | does anyone know what this error means |
| 09:59.10 | andycol | <PROTECTED> |
| 10:03.08 | MaliutaLap | that you have something set wrong in your dialplan? |
| 10:04.26 | MaliutaLap | anyone seen cisco 7960g handsets not getting dhcp leases when attached to a cisco switch? My server is seeing the discover packets and not sending a response, this is only happening on the handsets - other devices get leases no problems |
| 10:04.51 | andycol | can that cause a call to drop though? |
| 10:04.53 | andycol | or quality issues? |
| 10:05.49 | MaliutaLap | andycol: seriously I am willing to bet it's coming from a part of your dialplan where you have dial(SIP/${VAR}); |
| 10:06.19 | MaliutaLap | andycol: it would if a call is trying to execute that, and the variable is empty |
| 10:06.49 | MaliutaLap | andycol: pastebin your dialplan for me? |
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| 10:32.41 | andycol | let me pastebin it to you |
| 10:33.42 | andycol | its for this macro that they get the error |
| 10:33.43 | andycol | http://pastebin.com/WAHeWDTQ |
| 10:37.16 | andycol | MaliutaLap u still here? |
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| 10:38.01 | skirmisha | guys |
| 10:38.06 | MaliutaLap | yeah, just trying to fix a dhcp issue |
| 10:38.14 | andycol | np |
| 10:38.53 | skirmisha | anyone that knows how to make asterisk working with 2 ips on same interface? I need to replace src ip on outgoing packets, but i am not able to match them against incoming |
| 10:39.08 | andycol | yes make a virtual interface something like eth0:0 |
| 10:39.17 | skirmisha | yes i did that |
| 10:39.21 | zamba | that's just for aliasing.. you have to do policy-based routing |
| 10:39.31 | zamba | if you're going to use different gateways for the different ips |
| 10:39.46 | MaliutaLap | andycol: exten => s-AutoMagic,n,Dial(SIP/Automagic/${DIALED},tT) - I think you're setting the "DIALED" variable wrong |
| 10:39.47 | skirmisha | i do that, but i can't see output packet matched |
| 10:40.01 | andycol | MaliutaLap can i pm u? |
| 10:40.03 | skirmisha | its same gw just extra ip from same subnet |
| 10:40.22 | skirmisha | 2 days i am fighting with this and can't get it working |
| 10:40.36 | MaliutaLap | andycol: sure |
| 10:40.48 | skirmisha | i am marking packets with iptables , incoming, but outgoing do not match with incoming thus src ip is not changed |
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| 10:41.28 | MaliutaLap | skirmisha: you would need to have different "localnet" and "externip" for each lan in the right places |
| 10:41.45 | skirmisha | its same localnet |
| 10:41.54 | skirmisha | its extra ip from same subnet and same gw is used |
| 10:42.05 | MaliutaLap | I do it with a vlan on eth0 ... but then one IP is real, and one is private |
| 10:42.30 | skirmisha | i just have alias on eth0 |
| 10:42.41 | skirmisha | then i mark input on dst ip which is the alias ip |
| 10:42.44 | skirmisha | mark is ok |
| 10:43.00 | skirmisha | packets hit asterisk and then output is generated with no mark |
| 10:43.19 | skirmisha | even i try to restore mark its not picking up the session of the incoming packet |
| 10:43.40 | skirmisha | 2 days i am fighting with this and can't get it working |
| 10:44.38 | skirmisha | all the examples i saw they mark outgoing packets based on something, either port or src ip |
| 10:44.48 | skirmisha | but how would i match them with the session |
| 10:46.15 | skirmisha | any ideas ? |
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| 11:29.44 | WIMPy | Lovely how some multilingual telco announcements say completely different things. |
| 11:30.22 | Chainsaw | "I am out of the office, please send any materials to be translated to the other guy"? |
| 11:30.28 | Chainsaw | (In Welsh, of course) |
| 11:34.28 | TSM2 | is there still a bug in the ExecIf command thinking you have pipes as the delimiter when actually you are using a pipe as part of the exec command to run? |
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| 14:03.11 | MarkSx | Hey all, trying to use app_alarmreceiver, The call establishes but I only hear half of the first 1400hz tone then silence for the rest of the call. Any ideas ? |
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| 14:57.40 | _omer | https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure <------ canreinvite is not there in new realtime table structure ... Is it replaced by directmedia ? |
| 14:58.46 | Gugge | yes |
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| 15:00.48 | Gugge | it follows the options in sip.conf |
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| 15:02.15 | _omer | you mean global setting? |
| 15:02.40 | _omer | I got it ..... directmedia = canreinvite |
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| 15:39.49 | ChkDigit | Is there anything like a dialplan lint? |
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| 15:41.10 | [TK]D-Fender | ChkDigit: No, after we deplolyed res_dryer.so all of our lint is collected outside the system |
| 15:41.27 | kafal | Hi, I am facing one strange issue and needed some help. I am connecting to AMI using C# code. I am able to authenticate but after some time I get an error message that connection closed by remote host. I am pinging asterisk after authentication. any pointers? |
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| 15:58.28 | dan_j | Hi, Is directed call pickup possible with asterisk? |
| 16:00.23 | *** join/#asterisk DataWraith (~s1gny@188-192-189-82-dynip.superkabel.de) |
| 16:05.10 | [TK]D-Fender | dan_j: https://www.google.ca/#q=asterisk+directed+call+pickup |
| 16:10.12 | mjordan | kafal: Generally, the Asterisk log will tell you when it disconnected an AMI connection. It may do that for a whole host of reasons. |
| 16:20.19 | dan_j | When I try Pickup(), i get the following error |
| 16:20.20 | dan_j | http://pastebin.com/y427GfEE |
| 16:21.02 | dan_j | Whats SRTP module used for? |
| 16:22.04 | mbowie | is going to guess... Secure RTP? |
| 16:22.36 | dan_j | Any ideas why it can't find the channel? The ringing peer is SIP/201@sassons_phones |
| 16:25.33 | mjordan | that isn't what your log shows. Your log doesn't show what the name of the ringing channel is. |
| 16:25.52 | dan_j | Ok. one moment. I'll do it again. |
| 16:26.35 | dan_j | Actually, you are correct. Its wrong. |
| 16:26.40 | [TK]D-Fender | dan_j: SIP/201@sassons_phones <- that is not q valid channel |
| 16:26.45 | [TK]D-Fender | a* |
| 16:27.07 | [TK]D-Fender | dan_j: SIP/sassons_205-0000bf08 <-- this is what an * channel looks like... |
| 16:27.53 | dan_j | hang on. i need to supply the full channel name, or just the peer name? |
| 16:28.49 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Pickup |
| 16:28.55 | dan_j | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Pickup seems to imply its the peer name |
| 16:28.59 | mjordan | no it doesn't |
| 16:29.03 | dan_j | Pickup(extension |
| 16:29.09 | mjordan | Extension != Peer |
| 16:29.15 | dan_j | ok. understood. |
| 16:29.27 | dan_j | whats the simplest way of finding out the channel for the ringing peer? |
| 16:29.34 | mjordan | extension in Asterisk is a specific location in the dialplan. |
| 16:29.57 | ChkDigit | [TK]D-Fender - thanks for the tip on res_dryer.so, I'll make sure to plumb it in right. =) |
| 16:30.25 | mjordan | you can either setup pickup groups (or use PICKUPMARK), or you can tell it to pick up the ringing channel in a specific extension[@context] |
| 16:31.12 | dan_j | 'extension' being the extension in the dialplan that executed the Dial() command? |
| 16:31.22 | [TK]D-Fender | dan_j: "pickupchan" <- |
| 16:31.43 | dan_j | ah! |
| 16:34.28 | dan_j | Thank you |
| 16:34.34 | dan_j | Works perfectly |
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| 16:54.43 | pabelanger | no ice support in 1.8 right? |
| 16:54.55 | pabelanger | think 11 was the first version |
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| 17:44.03 | TazzNZ | "morning" all |
| 17:49.42 | mbowie | Moin TazzNZ |
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| 18:01.51 | pabelanger | okay, am I missing anything obvious |
| 18:01.52 | pabelanger | [2014-07-03 18:01:18.498] WARNING[1240][C-00000000]: chan_sip.c:10004 process_sdp: Insufficient information for SDP (m= not found) |
| 18:01.59 | pabelanger | http://pastebin.com/sejeGbru |
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| 18:19.34 | TazzNZ | pabelanger: I can't spot anything |
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| 18:23.30 | pabelanger | Hmm |
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| 18:39.28 | kafal | @mjordan: logs show the following message sometimes "ERROR[26662] utils.c: fwrite() returned error: Broken pipe" I Googled but couldn't find a fix. |
| 18:45.12 | mjordan | Most likely your client is not reading the events fast enough. |
| 18:45.26 | mjordan | If Asterisk takes too long to write to a socket, it bails. |
| 18:45.32 | mjordan | You can control the writetimeout in manager.conf |
| 18:45.42 | WIMPy | Or not at all? |
| 18:45.48 | mjordan | Or not at all :-) |
| 18:45.54 | mjordan | which will anger AMI greatly. |
| 18:46.13 | mbowie | makes a note not to anger teh AMI. |
| 18:46.16 | WIMPy | Doesn't stop people from doing it. |
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| 19:03.02 | kafal | Tried setting it to 1 min, but still no luck. Same settings work fine with asterisk 1.4 but not with 12x. Let me try increasing the value. |
| 19:05.43 | pabelanger | kafal, are you closing your socket while asterisk is still sending events? |
| 19:06.15 | [TK]D-Fender | AMI changed some things in 12.... |
| 19:08.53 | kafal | I am not closing the socket. I am using ping action to keep a persistent connection always open. |
| 19:09.14 | [TK]D-Fender | What "ping action"? |
| 19:09.15 | kafal | I login to AMI, then fire Agents action and start getting Agent events |
| 19:10.10 | kafal | this one https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerAction_Ping |
| 19:10.33 | kafal | in between I get a message that the remote host closed the connection |
| 19:11.01 | [TK]D-Fender | Something s being done out of order |
| 19:11.52 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 19:12.13 | kafal | The same code works in 1.4. I went through the change log of 12x but couldn't find anything which can result in connection being closed. |
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| 19:13.57 | kafal | my logger.conf has tis setting: full => notice,warning,error,debug,verbose,dtmf,fax |
| 19:14.18 | kafal | is there anything else that needs to be done to log more debug info? |
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| 19:36.59 | mjordan | debug information might yield more information leading up to the closing of the socket. |
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| 19:41.32 | kafal | @mjordan sorry my chat got dosconnected. I didn't get your full message. Can you please ping the message again? |
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| 19:43.51 | mjordan | <mjordan> debug information might yield more information leading up to the closing of the socket. |
| 19:52.29 | TazzNZ | does anyone here know of a web based solution to show who is currently in a conf. call, with talk highlighting ? |
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| 20:24.16 | derekb | hi there, having a strange issue, wondering if anyone has any ideas. |
| 20:24.22 | derekb | customer A makes a call to customer B |
| 20:24.38 | derekb | but the call gets rejected, for some reasons it seems liek theres a call loop happening |
| 20:24.48 | derekb | but if i change the outbound CLID of customer A, the call routes fine |
| 20:25.27 | Qwell | I have that happen too. |
| 20:25.30 | Qwell | Stupid customers always wanting to make calls. |
| 20:25.56 | mbowie | XD |
| 20:28.14 | derekb | ;) |
| 20:31.09 | TazzNZ | derekb: we need some console logs at least to start this |
| 20:31.11 | TazzNZ | ~pb |
| 20:31.12 | 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. |
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| 20:49.11 | qakhan | is there any calling card system with asterisk. |
| 20:49.28 | TazzNZ | qakhan: not built in |
| 20:49.34 | TazzNZ | you have to make your own |
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| 20:53.49 | qakhan | ok is there any link which descirbe which items required |
| 20:54.56 | TazzNZ | qakhan: you going to need more than that - you would need to code it |
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| 21:19.14 | _grim_ | can anyone tell me where to find the correct sql table structure for queue_members? |
| 21:20.34 | _grim_ | when agents are on the phone their status doesn't change. debug says: Device 'SIP/5001' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. But they clearly are as they are getting calls... |
| 21:25.57 | TazzNZ | _grim_: you'll have to give us more info |
| 21:26.03 | TazzNZ | what does your table look like at the moment ? |
| 21:26.14 | TazzNZ | how is your queue's setup |
| 21:26.17 | TazzNZ | etc |
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| 21:34.01 | _grim_ | TazzNz: http://pastebin.com/RJe2sWBR |
| 21:34.11 | wdoekes | _grim_: I think you're running into a limitation of realtime queues. |
| 21:34.55 | _grim_ | wdoekes: NOOOOOOOOOOOOOOOO don't say that ~ I'm already over budget on this project..... :'-( |
| 21:35.45 | _grim_ | if I remember correctly, realtime queues are the only way to create dynamic queues, correct? |
| 21:36.12 | wdoekes | unless you load queues from static realtime and 'queue reload' them when changes occur |
| 21:36.43 | _grim_ | :-P puke..... oh well... it is what it is... (maybe) (fingers crossed) |
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| 21:56.40 | _grim_ | this is kinda weird too.... hints always show idle.... http://pastebin.com/Nn2yCwGu |
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| 23:55.29 | ipengineer | Does anyone know how this would translate into the new PJSIP_HEADER function? SIPAddHeader,Alert-Info: info=alert-autoanswer |
| 23:56.47 | ipengineer | I tried this but for whatever reason it is not adding it. Doesnt look right to me Set(PJSIP_HEADER(add,Alert-Info:)=info=alert-autoanswer) |
| 23:57.22 | [TK]D-Fender | I'm pretty sure the ":" is bad |
| 23:57.47 | ipengineer | I tried without that but let me give it another go to make sure something else didnt change |
| 23:58.37 | ipengineer | That is the CLI output: https://gist.github.com/zconkle/a84a45103f73b173984f |