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| 01:14.43 | michael-i | Since I'm not coming up with anything on Le Google: Is it possible to initiate a Macro during a call via the AMI? I'd like to replace my applicationmap dtmf shortcut with an AMI call. |
| 01:15.54 | *** join/#asterisk vbman2 (~WildSide@ool-18b89939.dyn.optonline.net) |
| 01:16.08 | vbman2 | what kind of system would i need for 200 concurrent calls and 1,000 registrations |
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| 01:17.33 | WIMPy | michael-i: You can redirect any call to any place in the dialplan at any time or event. |
| 01:18.28 | michael-i | WIMPy: and the Event name is? Maybe I'm missing something obvious... |
| 01:19.27 | WIMPy | I have never used "features". Just connect to ami and look what's coming in that case. |
| 01:19.58 | michael-i | I need to send an event though, not look for one coming in. |
| 01:20.30 | WIMPy | No, you receive events. You send commands. |
| 01:21.08 | WIMPy | And if you don't wait for an event, but know when you want to do something, even easier. Just do it. |
| 01:21.21 | michael-i | arghâ¦yes. Sorry, the thing I'm programming sends these as events. |
| 01:21.21 | [TK]D-Fender | michael-i: How would you be calling AMI in the first place? |
| 01:21.59 | WIMPy | Call AMI? |
| 01:22.27 | WIMPy | Like ringing on its port? |
| 01:22.41 | michael-i | I'm just looking for the command(s) to send to the AMI to initiate recording on a bridged channel. |
| 01:23.25 | [TK]D-Fender | michael-i: Any reason standard on-demand monitor doesn't cut it? |
| 01:23.25 | michael-i | If I could just trigger the applicationmap entry I've already defined, that'd be cool. |
| 01:23.26 | WIMPy | 'monitor' |
| 01:24.02 | [TK]D-Fender | michael-i: IF you already have an applicationmap... why isn't that usable by itself? |
| 01:24.14 | WIMPy | You don't have to. You can control that directly via AMI. |
| 01:24.23 | raker | there is a current bug for asterisk that prevents from passing callerid prefixes when using followme |
| 01:24.33 | raker | is there any workarounds that people can suggest? |
| 01:24.39 | WIMPy | 'manager show commands' - See *monitor. |
| 01:24.47 | michael-i | [TK]D-Fender: monitor will probably work, I'm just doing some extra stuff in the macro which would be nice (recording count, recording owner, etc) |
| 01:25.17 | michael-i | reason for AMI vs appmap: don't want to use dtmf combos at all |
| 01:25.18 | WIMPy | Might be easier from your script than in the dialplan. |
| 01:25.30 | [TK]D-Fender | <PROTECTED> |
| 01:25.46 | [TK]D-Fender | I recall there being mention of some limitations on that |
| 01:26.21 | michael-i | [TK]D-Fender: I'm calling a Macro from the applicationmap which does all of these things. Works wonderfully, just can't have dtmf combos |
| 01:26.43 | [TK]D-Fender | michael-i: Meaning? |
| 01:26.54 | michael-i | ? |
| 01:28.38 | [TK]D-Fender | "DTMF combos". Not sure what you mean by this |
| 01:28.59 | michael-i | key presses, I don't want anyone to have to memorize key combinations |
| 01:29.18 | [TK]D-Fender | michael-i: Ok.. how would you be triggering this then? |
| 01:29.34 | michael-i | [TK]D-Fender: That's pretty much my question... |
| 01:29.48 | f2knight | have a tricky nat issue that is stumping me |
| 01:30.05 | WIMPy | michael-i: You should have asked that. |
| 01:30.09 | f2knight | 3 sip phones all on different networks and all behind nat |
| 01:30.15 | WIMPy | That obviousely depends on your phones. |
| 01:30.32 | f2knight | public asterisk box |
| 01:30.40 | f2knight | did's sent to asterisk box rings |
| 01:31.02 | f2knight | however audio to 1 location does not work. |
| 01:31.11 | [TK]D-Fender | michael-i: Ok, your beign unsure of how you want it to work differently doesn't help... |
| 01:31.19 | f2knight | audio to the other locations do work |
| 01:31.49 | f2knight | 1 device behind each router. all routers are SG580's and setup the same |
| 01:32.15 | michael-i | [TK]D-Fender: No, but that is my question which I asked initially: "Is it possible to initiate a Macro during a call via the AMI? I'd like to replace my applicationmap dtmf shortcut with an AMI call." |
| 01:32.21 | f2knight | dialing the users |
| 01:32.31 | f2knight | extension from any other works just find. |
| 01:33.16 | WIMPy | michael-i: No, you can't call a macro, but you can put a call into any placa of your dialplan, or for your task just start monitoring on a channel. |
| 01:34.31 | michael-i | WIMPy: â¦and that's where I asked you which command will do that jumping/goto? |
| 01:34.50 | WIMPy | 'redirect' |
| 01:35.26 | WIMPy | But if you do that you will break the bridge as you will terminate Dial(). |
| 01:35.50 | WIMPy | So better look at 'monitor', possibly alongside other options. |
| 01:36.18 | michael-i | WIMPy: ah, ok. Yeah, that will mess things up. Thanks though |
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| 01:36.50 | WIMPy | Well, I guess you could redirect both call legs and bridge them again after you're done without noticable interruption. |
| 01:36.59 | michael-i | monitor looks like the best betâ¦seems like a natural command for the AMI though: "execute this applicationmap entry on this channel" |
| 01:37.13 | [TK]D-Fender | michael-i: And how would you be triggering the AMI to start this off? You haven't quite claified the absolute initial trigger and what cascade you had in mind |
| 01:37.14 | WIMPy | But I'd recomend 'monitor'. |
| 01:38.05 | michael-i | [TK]D-Fender: There's a user portal web interface that the phones are associated with. I want the user to be able to click 'record' in there while in a call. |
| 01:40.16 | [TK]D-Fender | michael-i: Ok, so calls would be well identified.. then you could probably do it all via AMI withthe "Runn command against channel" option I believe I saw in the docs. |
| 01:40.36 | [TK]D-Fender | michael-i: Call recording is a very simple direct separate AMI command as well. |
| 01:41.57 | michael-i | [TK]D-Fender: It looks like I can do 90% of what I want with 'monitor.' But, the filename generation is kind of complicated and I wouldn't have enough info in the user portal to accomplish that via the command. Really need to be "in" asterisk to generate that filename. |
| 01:42.24 | michael-i | Could just rethink some other thingsâ¦this isn't a huge priority right now. Just fishing for info. |
| 01:42.42 | [TK]D-Fender | michael-i: What parts would you need from *? |
| 01:42.52 | [TK]D-Fender | michael-i: All the channel vars you can pull via AMI as it is... |
| 01:44.04 | michael-i | [TK]D-Fender: It is a bit more involved with that due to security concerns. This code runs in the browser via websockets. The user is then free to manipulate those variables. If they are read from * directly and never leave that scope, I'm golden. |
| 01:44.13 | michael-i | s/with/than |
| 01:44.21 | vbman2 | what specs do i need for 200 concurrent calls and 1,000 registrations |
| 01:44.56 | [TK]D-Fender | michael-i: Ok, you have separation concerns.... Keep in mind if you're running AMI... that is already a complete red-flag as it is. You can do jsut about anything anyway, and it's plain-text |
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| 01:45.33 | [TK]D-Fender | michael-i: But an idea might be to Oriignate a local channel, have that do the dirty work, and push back the result via a custom AMI event |
| 01:46.15 | michael-i | [TK]D-Fender: that sounds promising |
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| 01:46.50 | michael-i | [TK]D-Fender: we're looking at how to lock down AMI access on a context levelâ¦it's the scary monster in the closet |
| 01:47.54 | [TK]D-Fender | michael-i: I prefer the term "Free-agent Gremlin" :p |
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| 01:48.37 | michael-i | [TK]D-Fender: :) it is VERY daunting but totally justifiable for the user experience it enables |
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| 01:51.35 | rainkid | Somehow, I magically got automixmon to record in wav earlier. Now, it magically broke. TOUCH_MIXMONITOR_FORMAT and TOUCH_MONITOR_FORMAT both set to wav |
| 01:52.18 | michael-i | brb |
| 01:53.35 | rainkid | Anyone can help? Spending about 5 hours trying to figure out how to chang automixmon format isn't fun =( |
| 01:57.30 | f2knight | Q: Any one want to tackle a NAT audio issue? On site and able to replicate issue. |
| 01:57.59 | f2knight | inbound calls over DID do not transmit audio, if you put the caller on hold then pick up the calls connect. |
| 02:00.21 | rainkid | how about outbound? |
| 02:00.48 | f2knight | outbound works fine |
| 02:00.59 | rainkid | does the asterisk server have a public or private IP? |
| 02:01.03 | f2knight | public |
| 02:01.49 | rainkid | is the no-audio both direction? |
| 02:02.07 | f2knight | rainkid, yes |
| 02:03.13 | rainkid | if NAT is set up properly, i'll say check your firewall(s) |
| 02:03.42 | rainkid | oh, but that wouldn't explain why it would work after Hold... |
| 02:04.06 | f2knight | exactly. |
| 02:04.51 | f2knight | and it wouldn't explain why when dialing the extension and not the did I can get audio.. |
| 02:05.07 | f2knight | I am wondering if its possibly a dialplan flaw that I am overlooking. |
| 02:05.32 | rainkid | when you're dialing the extension you're internal. DID is external, correct? |
| 02:05.41 | f2knight | not exactly.. |
| 02:06.19 | f2knight | SIP Phones are all Natted on there own networks. they don not share the same network as each other. |
| 02:07.33 | f2knight | so if softphone1 dials the public DID and the dialplan routes it to where it is supposed to go , (softphone2) which is behind another nat |
| 02:07.43 | f2knight | nat -- public -- nat |
| 02:07.47 | rainkid | Ah. |
| 02:07.53 | f2knight | BUT!!! |
| 02:08.00 | rainkid | is this a new install? |
| 02:08.02 | f2knight | cell -- public -- nat does the same thing |
| 02:08.08 | f2knight | so its really any outside line. |
| 02:08.47 | f2knight | however if softphone1 calls softphone2 by there privateexten, then it connects fine.. which is still nat -- public -- nat |
| 02:09.14 | f2knight | I can tear this down to a basic dialplan if needed. just prefered not too |
| 02:09.20 | ChannelZ | so * is a public IP and not firewalled |
| 02:10.56 | f2knight | correct |
| 02:10.57 | rainkid | the only similar issue i've had is a single softphone software not communicating with my hardphones through nat. i attributed that to crappy softphone. |
| 02:11.09 | f2knight | * is naked to the world so to speak |
| 02:11.40 | ChannelZ | So it seems like more of an endpoint issue. is * aware of which peers are behind nat? Have you looked at a SIP debug or RTP debug to see where it thinks it's sending media to? |
| 02:11.41 | f2knight | rainkid, these are all hardphones |
| 02:11.51 | rainkid | hm |
| 02:12.12 | f2knight | ChannelZ, if you would be so kind as to tell me how to or what to look at I would love to do just that |
| 02:12.56 | f2knight | I have verbose and debug set to 10 |
| 02:13.05 | ChannelZ | sip set debug on |
| 02:13.25 | f2knight | is that why debug never displayed anything |
| 02:13.30 | ChannelZ | no |
| 02:14.12 | ChannelZ | sip debug is separate from "normal" debug (which in most people's cases is unnecessary) |
| 02:14.22 | f2knight | ahhh |
| 02:14.39 | f2knight | so core set debug 10 was not the right command |
| 02:15.24 | ChannelZ | yes and no |
| 02:15.55 | f2knight | not the right one for what I needed.. right now.. Okay so I got this running and lots of stuff is flying by. |
| 02:16.16 | ChannelZ | debug won't normally go to the console unless you configure it to; It probably filled up one of the disk logfiles |
| 02:16.59 | WIMPy | michael-i: You can get channel variables via AMI if that helps. |
| 02:17.09 | f2knight | Okay I see an notice that says ... |
| 02:17.13 | f2knight | let me pbin it |
| 02:17.37 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 02:18.33 | f2knight | http://pastebin.com/cBCg19AP |
| 02:18.53 | f2knight | that just jumped out at me. the UDP Blocked ... |
| 02:19.35 | ChannelZ | that's something specific to the grandstream, no idea what it takes that to mean |
| 02:20.26 | *** join/#asterisk coppice (~chatzilla@m121-202-96-32.smartone-vodafone.com) |
| 02:20.32 | ChannelZ | is that peer set to nat=yes in your sip.conf ? |
| 02:21.00 | ChannelZ | (you said they were all actually behind NAT) |
| 02:22.57 | f2knight | 15614477741103/1561447774 98.246.xx.xx D N A 5066 OK (136 ms) Cached RT |
| 02:23.52 | ChannelZ | ok so chances are Asterisk is sending its audio to the right place, but it's getting stopped on that end. |
| 02:24.13 | f2knight | ChannelZ, you think the router? |
| 02:24.27 | ChannelZ | Yes |
| 02:24.31 | f2knight | because heres whats odd |
| 02:24.40 | f2knight | the Grandstream phone has 4 lines. |
| 02:24.57 | ChannelZ | It's probably not mapping the incoming audio stream back to the phone through the firewall |
| 02:24.59 | f2knight | line 1 comes from another server but rings in fine |
| 02:25.05 | f2knight | and with audio |
| 02:25.31 | f2knight | this account is registered to line 4 |
| 02:25.38 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 02:26.33 | *** join/#asterisk adolfomaltez (~taro@190.62.230.28) |
| 02:26.33 | f2knight | and yes the database has nat=yes.. though i didn't see it in sip show peer |
| 02:27.15 | ChannelZ | It could be port related if the router/firewall it's behind is stateful and actually noticing the SIP traffic to do the port mapping correctly. It might not be seeing your line 4 because of the port 5066 (though I would think it'd be looking where the packet is going TO, not where it's coming from) |
| 02:27.23 | SeRi | [TK]D-Fender, You in? |
| 02:28.25 | *** join/#asterisk SBTech (~Erin@173-160-148-149-Washington.hfc.comcastbusiness.net) |
| 02:30.11 | f2knight | ChannelZ, if that was so then I should not even get the sip signialing. |
| 02:30.34 | ChannelZ | no |
| 02:30.42 | f2knight | but I am. and remember if I put it on hold then pick it up audio works again |
| 02:31.00 | ChannelZ | The SIP and the audio are two different connections |
| 02:31.11 | f2knight | I am not saying its not a nat issue.. just how to isolate and resolve is what I am looking to find. |
| 02:31.18 | f2knight | right audio is rtp stream |
| 02:31.38 | [TK]D-Fender | yes |
| 02:31.54 | ChannelZ | The hold/pickup is an oddity but could be explained. It's complicated. |
| 02:32.09 | michael-i | taking off. Thanks WIMPy and [TK]D-Fender for the feedback |
| 02:32.20 | f2knight | okay , oddity i agree. |
| 02:32.25 | [TK]D-Fender | No, that's because it forces a reinvite which is a very telltale sign that you didn't prevent them in the first place likew you're supposed to |
| 02:32.38 | [TK]D-Fender | And sip show peers doesn't prove where the RTP was negotiated |
| 02:32.51 | [TK]D-Fender | <PROTECTED> |
| 02:33.03 | f2knight | okay so how can we isolate it |
| 02:33.11 | ChannelZ | When you make a call, Asterisk asks the phone to send its audio to its IP on a certain port. The phone requests the same of Asterisk. However in neither case is it guaranteed a connection can actually be made. |
| 02:33.19 | [TK]D-Fender | Go look at a call, go show configs |
| 02:34.46 | ChannelZ | When you're behind NAT, the firewall/router has to be aware that you're going to receive an *incoming* connection so that it can send it to the right IP on the LAN side of the network. This either has to be statically mapped, or can be mapped automatically by the router/firewall in a couple different ways (which is the part that is hard to know what your setup is or isn't doing.) |
| 02:34.47 | f2knight | [TK]D-Fender, that does not return anything, * 1.8 here |
| 02:35.20 | [TK]D-Fender | "What" doesn't retun anything? |
| 02:35.49 | ChannelZ | You can look at the SIP debug after placing a call and see what each side is telling the other, to verify that's all correct. Assuming it is then you've got to figure out why the traffic isn't making it in or out of whatever side as the case may be |
| 02:35.55 | f2knight | voice1*CLI> core show config |
| 02:35.55 | f2knight | No such command 'core show config' (type 'core show help core show config' for other possible commands) |
| 02:35.55 | f2knight | voice1*CLI> core show configs |
| 02:35.55 | f2knight | No such command 'core show configs' (type 'core show help core show configs' for other possible commands) |
| 02:35.55 | f2knight | voice1*CLI> show configs |
| 02:35.56 | f2knight | No such command ' show configs' (type 'core show help show configs' for other possible commands) |
| 02:36.01 | [TK]D-Fender | .... |
| 02:36.14 | [TK]D-Fender | facepalms |
| 02:36.39 | rainkid | (so, anyone know how to change automixmon file format?) |
| 02:36.56 | rainkid | is about to give up |
| 02:37.34 | ChannelZ | rainkid: I have no idea, my guess is that it records in whatever the 'current' channel format is |
| 02:37.38 | f2knight | ChannelZ, okay so before I go figure out how to capture the sip debug, if I have line 1 on the phone.. oport 5060 working with in bound calls audio the works.. then asuming I disable that line and reasign the issue line to the port 5060 , audio 'should' work. |
| 02:37.53 | ChannelZ | No, it "might" work |
| 02:38.27 | rainkid | ChannelZ: Oooh maybe that's why it kinda worked earlier but now it doesn't. |
| 02:38.33 | f2knight | ChannelZ, but it would be a good guess? inother words should I try it before digging in to all the sip headers.. (they confuse me) |
| 02:39.02 | ChannelZ | If the router the phone is behind is doing packet inspection and trying to setup the incoming NAT routing for certain protocols, it's possible it looks at SIP and is doing magic for you.. and that it's not working in this case because of the port number. It's just one of 10 different possibilities |
| 02:40.01 | f2knight | ChannelZ, okay so that sounds like its worth a quick try to find out. 2 of the locations have the same router,,, and both have the same issue |
| 02:40.13 | f2knight | only one has it with port 5060... . umm |
| 02:40.57 | ChannelZ | like I said it's probably unlikely, only because I would expect the router to be paying attention to where the packet is going TO, not where it's coming FROM. I'm only guessing. |
| 02:41.21 | ChannelZ | (IE it's still being sent TO 5060 on your Asterisk server) |
| 02:41.29 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca) |
| 02:42.32 | ChannelZ | rainkid: actually I think there are some channel variables you can set... looking |
| 02:43.06 | f2knight | okay i just did tcpdump -s 2000 -w mycapp-.pcap port 5060 or portrange 10000-20000 |
| 02:43.24 | rainkid | I tried setting TOUCH_MONITOR_FORMAT and TOUCH_MIXMONITOR_FORMAT to no avail. |
| 02:43.31 | f2knight | but I really do not know what to do in wireshark to analyse it.. |
| 02:46.49 | ChannelZ | well I think Wireshark recognizes SIP traffic and will break out the conversation |
| 02:47.13 | ChannelZ | rainkid: hmm yeah those are the ones though I'm not sure where they are documented what you should set them to |
| 02:48.03 | f2knight | ChannelZ, yes but I am not sure what I am looking at. |
| 02:48.06 | rainkid | Yes. I've been looking for documentation as well, but it does not exist. I don't even know if TOUCH_MIXMONITOR_FORMAT is valid or not. And.. sadly, C code isn't my specialty. |
| 02:48.59 | ChannelZ | what have you tried setting it to? |
| 02:49.05 | rainkid | wav |
| 02:49.22 | ChannelZ | looks like that should be default anyway |
| 02:49.28 | rainkid | it defaults to gsm. |
| 02:49.42 | ChannelZ | what version of Asterisk? |
| 02:49.43 | rainkid | thought some documentation states that it defaults to g729 |
| 02:49.46 | rainkid | 1.8.5 |
| 02:50.16 | ChannelZ | hmm is that what format the call is in? |
| 02:51.04 | rainkid | (shhhh.. how do I check?) |
| 02:51.13 | ChannelZ | Just browsing through the source it seems to imply it defaults to wav if not specified but I'm not familiar with the source at all so something else might be going on I've just not seen yet the way things are stitched together |
| 02:51.32 | dijib | how does my dialplan look to Dial() an iNum number |
| 02:51.34 | rainkid | i did see mention of TOUCH_MIXMONITOR_FORMAT in feature.c |
| 02:51.42 | f2knight | rainkid, did you look here https://wiki.asterisk.org/wiki/display/AST/Various+application+variables |
| 02:52.00 | ChannelZ | f2knight: they are listed there but not really defined |
| 02:52.07 | f2knight | its not TOUCH_MIXMONITOR_ it is TOUCH_MONITOR_ |
| 02:52.10 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
| 02:52.23 | rainkid | TOUCH_MONITOR is for automon |
| 02:52.25 | ChannelZ | It's actually both. |
| 02:52.26 | rainkid | I'm using automixmon |
| 02:52.37 | f2knight | oh .. sorry . was trying :( |
| 02:52.42 | rainkid | Thanks =) |
| 02:52.54 | rainkid | has been trying to figure this out for 7 hours now |
| 02:53.19 | f2knight | so i have this pcap open in ws,,, just what am i looking for? |
| 02:53.42 | ChannelZ | rainkid: you're doing this with a feature code right? |
| 02:53.53 | rainkid | yes |
| 02:54.12 | f2knight | Wait... SIP/SDP? |
| 02:54.38 | ChannelZ | SIP |
| 02:55.17 | rainkid | How can I check what format a call is in from CLI? |
| 02:55.21 | f2knight | I am seeing it say SIP/SDP...what excatly is that ? |
| 02:55.59 | rainkid | "SDP is a description protocol, SDP messages can be transported by means of different protocols, for example SIP." |
| 02:56.05 | ChannelZ | SDP is like a sub-protocol of describing the audio session |
| 02:56.30 | f2knight | okay but should it be being used? or could it be causing an issue? |
| 02:56.47 | ChannelZ | if it wasn't being used you'd have no audio for sure |
| 02:56.55 | f2knight | oh ok |
| 02:57.02 | ChannelZ | cuz neither side would know where to send it, what flavor to send it in... |
| 02:58.14 | f2knight | i am trying to go thorugh this .. so I know that my line is on 5066 |
| 02:58.48 | ChannelZ | well, it's 'sending from' port 5066 |
| 03:00.10 | ChannelZ | rainkid: sorry from earlier -- while in a call, "sip show channels" should tell you the format |
| 03:00.47 | dijib | how do i dial this number? +883 5100 0000 0093 |
| 03:00.49 | rainkid | ulaw format. automixmon saving as .raw file. |
| 03:01.44 | ChannelZ | but you say it's actually gsm data? |
| 03:01.50 | dijib | just put the .ulaw extension in the mixmonitor filename |
| 03:02.19 | rainkid | that was an assumption. i have a gsm autoplayer that can play the raw file. |
| 03:03.24 | f2knight | rainkid, if i remember right.. doesn't trixbox/freepbx call a shell script that uses sox to convert it ? |
| 03:04.11 | rainkid | f2knight: automon saves as two separate audio channels. automixmon saves as one - thus, the reason for me using it. |
| 03:04.48 | f2knight | rainkid, yes, but I do recall that freepbx saves as a single file with both legs. and as a wav. |
| 03:05.51 | ChannelZ | dijib: not sure if you have to dial extra country codes or how all that mess works.. is that what you're asking? |
| 03:06.24 | dijib | i figured it out, its covered by the _011. |
| 03:06.27 | dijib | extension |
| 03:06.52 | dijib | its a iNum number... free IP to IP calling |
| 03:07.39 | f2knight | ChannelZ, okay ... just tested and no moving the line to the 5060 does not solve the issue. |
| 03:07.56 | f2knight | I am thinking it is something with the way the account is set |
| 03:08.25 | ChannelZ | so can you pastebin the sip debug showing the whole call? |
| 03:08.44 | f2knight | can I get that out of the pcap? |
| 03:08.50 | ChannelZ | (hint: it will be half a dozen messages flying back and forth) |
| 03:09.06 | ChannelZ | well I'd assume wireshark parsed that for you... |
| 03:09.17 | ChannelZ | that's kind of its point.. |
| 03:09.22 | f2knight | ChannelZ, do you know how to use wireshark? |
| 03:10.03 | ChannelZ | not really, I haven't used it in some time |
| 03:10.24 | f2knight | okay because I do not knwo how to save the output lol |
| 03:12.01 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 03:14.01 | rainkid | So.. I finally figured out what's happening. Asterisk -> External, format is ulaw, saves as RAW. External -> Asterisk, format is ulaw, saves as WAV. |
| 03:14.13 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 03:14.14 | rainkid | Yes, more data that makes no sense! |
| 03:18.13 | ChannelZ | when you say "saves as raw" and "saves as wav" are you meaning the file is called xxx.raw or xxx.wav or how are you determining the actual data type? |
| 03:18.29 | f2knight | ChannelZ, can i do this just for the one extenision? the debug that is |
| 03:18.38 | rainkid | Yes, they are saved with those extensions. |
| 03:18.41 | ChannelZ | sip set debug ip x.x.x.x |
| 03:19.04 | f2knight | ChannelZ, IP of the ... sip phone ? |
| 03:19.13 | rainkid | The wav files play fine. Googling reveals that the raw files are probably headerless ulaw audio. |
| 03:19.14 | ChannelZ | yes |
| 03:19.19 | f2knight | ChannelZ, the public ip of said sip phone i mean |
| 03:20.39 | ChannelZ | rainkid: and if you Set(TOUCH_MIXMONITOR_FORMAT=wav) prior to your Dial, you still get a .raw file? |
| 03:21.21 | rainkid | ChannelZ, I get it [globals]ly, and also tried in extensions.ael. Let me try it prio to the dial. |
| 03:21.32 | rainkid | s/get/set |
| 03:23.37 | f2knight | ChannelZ, okay want me to post the whole thing? |
| 03:23.49 | rainkid | exten => _1NXXNXXXXXX,3,Set(TOUCH_MIXMONITOR_FORMAT=wav) before my Dial results in RAW file. |
| 03:23.58 | ChannelZ | f2knight: I guess |
| 03:25.53 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 03:26.28 | f2knight | http://pastebin.com/gsDqahgs |
| 03:26.41 | ChannelZ | rainkid: hmm I get .wav here |
| 03:27.09 | rainkid | From asterisk calling out? |
| 03:27.19 | ChannelZ | yes, calling my cell from a softphone |
| 03:27.48 | rainkid | Which version of Asterisk are you using? |
| 03:28.19 | ChannelZ | and if I set it to ulaw, I get ulaw |
| 03:28.26 | ChannelZ | 1.8.5.0 |
| 03:28.58 | rainkid | I just want to make sure I am not missing anything: features.conf:automixmon => ** |
| 03:29.16 | ChannelZ | I'm using *3 but it either starts or it doesn't |
| 03:29.16 | rainkid | and Set(TOUCH_MIXMONITOR_FORMAT=wav) prior to Dial line. |
| 03:29.19 | ChannelZ | I assume you see it on the console |
| 03:29.27 | rainkid | Yes. I tried the default of *3 too |
| 03:29.31 | rainkid | Yes. |
| 03:29.53 | rainkid | I am also running 1.8.5.0 |
| 03:30.04 | rainkid | * hates me =( |
| 03:30.26 | rainkid | Does it have anything to do with clearglobalvars? |
| 03:30.29 | ChannelZ | Are you sure you're messing with the right extension? Do you see it set the variable in the console and all that? |
| 03:31.13 | rainkid | Yes. When I set it globally, and I do a reload in CLI, it TOUCH_MIXMONITOR_FORMAT appears in 'setting global variable' |
| 03:31.17 | ChannelZ | not that I could think of no |
| 03:32.03 | rainkid | I only use one SIP trunk for outbound calls. I set it in that context. |
| 03:33.14 | rainkid | Do I need DYNAMIC_FEATURES = automixmon in globals? |
| 03:34.36 | ChannelZ | no |
| 03:35.44 | f2knight | ChannelZ, I need a smoke break this is driving me buggy. be back in about 20., maybe you will see something I don't |
| 03:36.59 | rainkid | Setting "TOUCH_MIXMONITOR_FORMAT=wav" under [globals] should work too, right? |
| 03:37.11 | ChannelZ | I would guess so but let me try |
| 03:38.26 | ChannelZ | seems so |
| 03:38.50 | rainkid | Can you just test another format so that I know it works 100% for you? |
| 03:39.49 | ChannelZ | yeah I've done both ulaw and wav |
| 03:40.25 | ChannelZ | I'm trying to see how I can even get mine to make a .raw |
| 03:40.38 | rainkid | set to ulaw |
| 03:40.50 | ChannelZ | well you said it was making a file called whatever.raw |
| 03:40.51 | ChannelZ | ? |
| 03:40.54 | rainkid | yes |
| 03:41.12 | ChannelZ | but you haven't set that anywhere |
| 03:41.14 | rainkid | i think it's a headerless ulaw audio file, despite the .raw extension |
| 03:41.18 | rainkid | Nope. I have not. |
| 03:41.27 | ChannelZ | that's what I"m trying to dupe cuz I've never seen that before |
| 03:42.35 | ChannelZ | left unset I get .wav as I'd though, judging by the source |
| 03:42.48 | rainkid | =( |
| 03:42.52 | ChannelZ | I think you maybe have something else in your dialplan going on you're not seeing or... I'm not really sure |
| 03:43.34 | rainkid | Well.. thanks for the help. At least you verified that it works for this version of *. |
| 03:43.48 | rainkid | I'll comb through everything slowly, perhaps try a new install |
| 03:43.51 | rainkid | and see what happens. |
| 03:44.01 | ChannelZ | is this packaged or from source? |
| 03:44.04 | rainkid | Source |
| 03:44.15 | ChannelZ | I can't imagine a bunk install would cause this behavior |
| 03:44.26 | ChannelZ | And it's plan Asterisk, not FreePBXmess ? |
| 03:44.32 | rainkid | Plain *. |
| 03:44.34 | ChannelZ | s/plan/plain/ |
| 03:44.43 | rainkid | I have no experience with any * GUIs. |
| 03:45.01 | rainkid | Hmm |
| 03:45.06 | rainkid | I do have some modules noload-ing. |
| 03:45.12 | rainkid | I wonder if that affects it. |
| 03:45.26 | ChannelZ | Put a NoOp(${TOUCH_MIXMONITOR_FORMAT}) prior to your Dial |
| 03:45.39 | rainkid | Sure. Let me try that. |
| 03:47.23 | rainkid | NoOping TOUCH_MIXMONITOR_FORMAT still resulted in .raw file |
| 03:47.41 | *** join/#asterisk Sakuranbo (~Sakuranbo@59.152.236.158) |
| 03:48.16 | ChannelZ | yeah but what did it say on the console? |
| 03:48.29 | Sakuranbo | Hi guys |
| 03:48.51 | ChannelZ | hi |
| 03:48.56 | rainkid | What am I suppose to look for? I just see the messages of MixMonitor beginning and ending |
| 03:49.21 | ChannelZ | core set verbose 3 |
| 03:49.27 | Sakuranbo | I am back |
| 03:49.50 | rainkid | Oh. I do see the "Executing [18002255288@outgoing-voicepulse:3] NoOp("SIP/100-00000000", "") in new stack" |
| 03:49.59 | rainkid | Um.. |
| 03:50.13 | ChannelZ | So that's not good. |
| 03:50.22 | ChannelZ | (I still don't know where it's getting "raw" from) |
| 03:50.47 | ChannelZ | pastebin your extensions.conf |
| 03:50.57 | rainkid | How did NoOp(${TOUCH_MIXMONITOR_FORMAT}) become NoOp("SIP/100-00000000", "")? |
| 03:51.28 | rainkid | Okay. Let me paste it. One moment |
| 03:52.01 | ChannelZ | The SIP/100-0000000 is the channel name it's executing on, "" is the contents |
| 03:52.08 | ChannelZ | which should be "wav" or whatever you set that variable to |
| 03:52.22 | ChannelZ | which means it's not set right. Type-o, or something else, not sure. |
| 03:53.06 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 03:54.26 | Sakuranbo | just wondering when would a call be "AUTO-destroyed" from the chan_sip.c script |
| 03:54.36 | rainkid | Sorry. I had a typo. Now it's "Executing [18002255288@outgoing-voicepulse:4] NoOp("SIP/100-00000000", "wav")" but still saving as .raw |
| 03:54.43 | rainkid | Let me paste my extensions.conf |
| 03:54.57 | dijib | your just trying to record outgoing calls? |
| 03:55.33 | rainkid | I'm trying to record calls on demand, and save as wave files for easy processing, instead of awkward ulaw files |
| 03:56.12 | dijib | on demand using *1 ? |
| 03:56.25 | rainkid | automixmon *3 |
| 03:56.57 | dijib | so u just put the r option in your dial command and make sure your feature.conf has it enabled |
| 03:57.19 | dijib | and make sure the application is installed |
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| 03:57.39 | ChannelZ | dijib: the problem isn't that it's not recording, it's that it's doing so in the undesired format |
| 03:57.46 | ChannelZ | and 'r' isn't the right option |
| 03:57.47 | Sakuranbo | I am tracing and debugging the script written by my predecessor, a Macro "mobile" has been written for making quick dial for dialing mobile number from the office internally |
| 03:58.46 | ChannelZ | rainkid: I see where 'raw' is coming from in the source now at least. Pondering a reason why it's happening for you |
| 03:59.34 | dijib | lets see that dialplan rainkid |
| 03:59.47 | Sakuranbo | the entire handshake and info passing were exactly the same except for one number I got the "sip_autodestruct" funciton called up |
| 04:00.04 | rainkid | http://pastebin.com/1LCn1TD1 |
| 04:00.22 | rainkid | It's not a very good dialplan =( |
| 04:03.41 | ChannelZ | Where are your recordings going to? |
| 04:04.23 | rainkid | default location /var/spool/asterisk/monitor |
| 04:04.39 | ChannelZ | wait.. is your phone in the 'outgoing' context? |
| 04:05.01 | rainkid | When I dial out, yes. |
| 04:05.26 | ChannelZ | I wonder if this is something screwy going on because of the goto |
| 04:06.03 | rainkid | I should rewrite without GoTos, right? |
| 04:06.13 | rainkid | This dialplan is from the 1.2 days, I think |
| 04:06.53 | ChannelZ | just for the hell of it, reset your sip peer to use the outgoing-voicepulse context directly and do the call again and see if it behaves any differently. I'm not thinking why it would but something odd is going on here |
| 04:07.10 | rainkid | Sure. Let me give it a shot |
| 04:07.14 | dijib | dont you want the ${TOUCH_MIXMONITOR_OUTPUT} variable? |
| 04:07.37 | rainkid | The variable is set. Just not.. used. |
| 04:07.46 | ChannelZ | It's set afterwards to tell you what filename it invented was |
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| 04:09.41 | rainkid | No go. |
| 04:10.05 | rainkid | I changed out outgoing context to just the outgoing-voicepulse |
| 04:10.13 | ChannelZ | and reloaded yes |
| 04:10.17 | rainkid | context directly |
| 04:10.29 | rainkid | Yes. I restarted asterisk completely just to be sure. |
| 04:10.45 | rainkid | file created is auto-1318392560-2129378997.raw |
| 04:11.17 | ChannelZ | ok. No surprise really. |
| 04:11.31 | rainkid | I keep getting this warning, but I think it's unrelated: " WARNING[30381]: app_mixmonitor.c:506 mixmonitor_exec: No volume level was provided for the heard volume ('v') option." |
| 04:11.43 | rainkid | Default volume is plenty loud for me. |
| 04:13.30 | rainkid | let me give it a shot with all modules loaded. |
| 04:14.06 | ChannelZ | So from the source, here's what i can tell you; When it actually starts the monitor, it looks for a / in the filename. Then it looks for a . in the filename and whether or not its location is later than the /. If it's not, then it sets the filename extension to .raw |
| 04:16.09 | rainkid | I wonder why the filename would be different for inbound and outbound calls. |
| 04:16.23 | rainkid | Assuming this is the reason why it's deciding to set it to raw. |
| 04:20.20 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 04:21.16 | rainkid | So.. SIP2SIP is wav. External to SIP is wav. Asterisk to External is raw |
| 04:24.05 | ChannelZ | do you still have the same source you originally built it from? |
| 04:24.14 | rainkid | yes |
| 04:24.33 | rainkid | Let me guess.. manually put a / in there? |
| 04:39.26 | f2knight | ChannelZ, back did you by chance peak at the sample? |
| 04:43.56 | ChannelZ | yeah I didn't see anything odd except for a couple of re-transmits, but I think everything was reporting all the right IPs (didn't see any LAN ones) so all I can really say is it's probably the router on the phone's side. Traffic isn't making it in. |
| 04:44.25 | ChannelZ | Why it does when you put it on hold and pick it back up I don't know, not sure I have a good theory on that one |
| 04:45.22 | f2knight | yea thats what I was afraid of |
| 04:47.09 | ChannelZ | is 98.246.74.16 the IP of the phone in question? |
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| 04:56.33 | f2knight | ChannelZ, thats the actuall public ip of the router but yes |
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| 05:27.24 | f2knight | ChannelZ, you still here? |
| 05:27.32 | f2knight | I think i might have partly tracked it down |
| 05:27.42 | f2knight | but need a little help defining it more |
| 05:31.02 | f2knight | I created a new account... and attached called it and it worked fine. |
| 05:31.57 | f2knight | so the only difference is that my accounts are in asterisk realtime. well the accounts I want are in asterisk realtime. |
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| 06:51.00 | vassilux | hi alls, I have dahdi with B410 board, sometime my span 3 go down and I can see in the full log file [Oct 11 01:00:15] VERBOSE[7334] logger.c: == Primary D-Channel on span 3 down |
| 06:51.00 | vassilux | [Oct 11 01:00:15] WARNING[7334] chan_dahdi.c: No D-channels available! Using Primary channel 9 as D-channel anyway! any idea ? |
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| 06:55.48 | vassilux | in my system.conf I have span=1,0,0,ccs,ami for span 1 is is correct for BRI ? |
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| 07:09.06 | kaldemar | vassilux: what is it connected to? |
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| 07:21.23 | zamba | anyone using a dinstar gsm gateway here? |
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| 07:43.18 | vassilux | it is connected to telco side ansd signalig bri_cpe |
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| 07:57.56 | PARAG1 | Hi guys, I am unable to find res_odbc.so in modules |
| 07:58.11 | PARAG1 | I compiled by "make All" |
| 07:59.14 | irroot | PARAG1 make menuconfig and ensure res_odbc is selected |
| 07:59.41 | PARAG1 | yes it is selected irroot |
| 07:59.55 | irroot | check modules.conf |
| 08:00.11 | PARAG1 | it is showing res_odbc = xxx |
| 08:00.13 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 08:00.21 | PARAG1 | yes modules.conf also |
| 08:00.23 | PARAG1 | it is enabled |
| 08:00.23 | kaldemar | PARAG1: which means that it is not selected. |
| 08:00.50 | kaldemar | PARAG1: you need to install the dependencies for it and re-run configure and recompile. |
| 08:00.53 | irroot | PARAG1 you need to have the odbc libs unixodbc and headers |
| 08:01.11 | PARAG1 | <PROTECTED> |
| 08:01.27 | PARAG1 | i already installed unixodbc and unixodbc-devel |
| 08:01.51 | PARAG1 | for odbc libs which package do i need to install |
| 08:02.01 | PARAG1 | mysql-odbc-connector i hv installed |
| 08:04.21 | PARAG1 | irroot: help pls |
| 08:05.39 | irroot | PARAG1 when you run configure check the config.log and see there what the problem is |
| 08:08.04 | PARAG1 | irroot: /usr/bin/ld: cannot find -liodbc |
| 08:08.59 | irroot | PARAG1 what system you using iodbc and unixodbc are 2 options did it look for unixodbc also ?? |
| 08:10.23 | PARAG1 | i m using unixodbc |
| 08:10.41 | PARAG1 | not sure |
| 08:10.49 | irroot | check in config.log |
| 08:10.49 | PARAG1 | if it look for unixodbc or not |
| 08:11.10 | PARAG1 | libodbc.so -> libodbc.so.2.0.0 in /usr/ib |
| 08:11.12 | PARAG1 | /usr/lib |
| 08:12.37 | PARAG1 | GENERIC_ODBC_INCLUDE='' |
| 08:12.38 | PARAG1 | GENERIC_ODBC_LIB='' |
| 08:12.38 | PARAG1 | UNIXODBC_DIR='' |
| 08:12.38 | PARAG1 | UNIXODBC_INCLUDE='' |
| 08:12.38 | PARAG1 | UNIXODBC_LIB='' |
| 08:12.44 | PARAG1 | all are empty |
| 08:12.54 | PARAG1 | not sure if i need to specify during ./configure |
| 08:13.41 | irroot | <PROTECTED> |
| 08:13.47 | irroot | look for the options for odbc |
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| 09:21.54 | WIMPy | irroot: Looks like your version number check is missing a 0 in all cases in the chan_lcr patch. |
| 09:22.20 | irroot | WIMPy mmm ill double check i copy pasted it from version.h |
| 09:22.46 | WIMPy | Current 1.8 has 999999. |
| 09:23.15 | WIMPy | I just tried to compile with that and it did the 10 version. |
| 09:23.16 | irroot | ok should have checked older versions |
| 09:25.02 | irroot | bangs head will fix |
| 09:25.27 | WIMPy | "Short copy"? ;-) |
| 09:25.32 | irroot | WIMPy have had a problem with lcr dying |
| 09:25.42 | irroot | better than a short something else :P |
| 09:25.48 | PARAG1 | irroot: hi i tried everything ---with-unixodbc, --with-odbc but nothing worked......I require res_odbc.so |
| 09:25.50 | PARAG1 | module |
| 09:25.50 | WIMPy | LOL |
| 09:26.14 | WIMPy | irroot: Any idea, when/how? |
| 09:26.51 | irroot | comes up on console spewing fire |
| 09:26.57 | irroot | ill dig into it latter |
| 09:27.33 | WIMPy | I do get the odd message sometimes. But it doesn;t do any harm for me. (except for the optics) |
| 09:27.51 | kaldemar | PARAG1: how about the ltld dependency? |
| 09:28.13 | irroot | WIMPy in this case need to kill lcr and restart it |
| 09:28.37 | PARAG1 | kaldemar: /usr/bin/ld: cannot find -liodbc |
| 09:28.49 | PARAG1 | which package do i need to install for ltld ? |
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| 09:29.34 | kaldemar | PARAG1: what system are you on? |
| 09:29.55 | PARAG1 | redhat 6.1 |
| 09:31.10 | kaldemar | PARAG1: maybe libtool-ltdl |
| 09:32.20 | PARAG1 | kaldemar: Package libtool-ltdl-2.2.6-15.5.el6.i686 already installed and latest version |
| 09:33.29 | *** join/#asterisk jkroon (~jkroon@dsl-241-244-137.telkomadsl.co.za) |
| 09:34.47 | kaldemar | PARAG1: how about libtool-ltdl-devel? |
| 09:38.59 | PARAG1 | its not present |
| 09:39.01 | PARAG1 | let me install |
| 09:41.40 | PARAG1 | kaldemar: still /usr/bin/ld: cannot find -liodbc |
| 09:42.14 | *** join/#asterisk darkskiez (~darkskiez@62-50-207-4.client.stsn.net) |
| 09:43.01 | PARAG1 | kaldemar: it is installed libtool-ltdl-devel-2.2.6-15.5.el6.i686 |
| 09:43.30 | PARAG1 | mysql-connector-odbc-5.1.5r1144-7.el6.i686 |
| 09:43.52 | PARAG1 | unixODBC-2.2.14-11.el6.i686 |
| 09:43.53 | PARAG1 | unixODBC-devel-2.2.14-11.el6.i686 |
| 09:43.58 | PARAG1 | all packages are their |
| 09:43.59 | PARAG1 | :( |
| 09:45.25 | kaldemar | make distclean && ./configure |
| 09:47.23 | irroot | PARAG1 if it can be built configure will pick it up |
| 09:47.32 | irroot | the errors are in config.log |
| 09:48.51 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 09:49.05 | irroot | mandla hi there long time |
| 09:49.51 | mandla | irroot, hey man, true, i managed to complete the Asterisk Testing, now im piloting it. |
| 09:50.02 | irroot | awesome |
| 09:50.34 | mandla | irroot, yah hey. |
| 09:50.51 | mandla | irroot, after so much straggle. |
| 09:50.58 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 09:51.06 | mandla | irroot, are you running linux?? |
| 09:51.10 | irroot | but you gained experiance |
| 09:51.16 | irroot | yip i do |
| 09:51.37 | mandla | irroot, gained a lot of experience my man. |
| 10:05.49 | *** join/#asterisk wonderworld (~ww@port-92-201-46-44.dynamic.qsc.de) |
| 10:08.21 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 10:12.55 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 10:13.20 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:13.39 | angryuser | Any sangoma guys here ? |
| 10:14.40 | mandla | angryuser, yah it me. |
| 10:14.45 | mandla | angryuser, lol |
| 10:15.03 | angryuser | mandla, Line Code Violation : 300 for E1 |
| 10:15.39 | angryuser | a101 E1 card, symptomps, when dialing out getting cause caode 28 |
| 10:15.55 | angryuser | incoming calls are ok, and i am sure of my dialplan |
| 10:16.25 | angryuser | Cause code means bad number format, however, i am sure it is a good one. |
| 10:16.37 | angryuser | mandla, lastes sangoma drivers |
| 10:16.40 | WIMPy | obviousely not. |
| 10:16.52 | WIMPy | Probably a wrong number type. |
| 10:17.21 | angryuser | WIMPy, Same setup, before the upgrade on asterisk 1.2 = No problems = Same dialplan |
| 10:17.41 | angryuser | WIMPy, So i am sure of my numbers, i even called 911 |
| 10:17.55 | WIMPy | same chan_dahdi.conf? |
| 10:18.16 | angryuser | WIMPy, zaptel > to dahdi changed, but pretty much the same |
| 10:18.25 | *** join/#asterisk xnfinite (~xnfinite@108.70.78.188.dynamic.jazztel.es) |
| 10:18.35 | angryuser | PRI CPE 31 channels, same group, |
| 10:19.00 | WIMPy | Check the dialplan options. |
| 10:19.03 | angryuser | mandla, your are not a tech i suppose ? |
| 10:19.17 | angryuser | WIMPy, What ? |
| 10:19.20 | mandla | angryuser, im not. |
| 10:19.58 | WIMPy | chan_dahdi.conf |
| 10:20.57 | angryuser | WIMPy, http://pastebin.ca/2089333 classics |
| 10:21.07 | angryuser | I repeat calls come in fine |
| 10:21.24 | WIMPy | It doesn't affect incumming calls. |
| 10:21.56 | WIMPy | Add pridialplan=unknown. I guess that's it. |
| 10:22.32 | angryuser | WIMPy, pri intense debug does show the return of cause code 28 however i am can real all the Qsiq 931 ;( |
| 10:22.34 | WIMPy | That should default to unknown, but I think it doesn't. |
| 10:22.37 | angryuser | WIMPy, lets try |
| 10:23.31 | WIMPy | or is it prilocaldialplan? |
| 10:23.55 | WIMPy | The documentation could definitely be better. |
| 10:24.55 | WIMPy | Well, best to set both. |
| 10:25.23 | angryuser | WIMPy, pridialplan=unknown worked, nice catch |
| 10:25.46 | WIMPy | So far for sensible defaults. |
| 10:25.53 | angryuser | WIMPy, i completely forgot about it |
| 10:26.22 | WIMPy | I guess you shouldn't have to think about that. |
| 10:26.26 | angryuser | WIMPy, i admin lately i was using a lot of patton gateways, lost my hands on exp a bit |
| 10:26.33 | angryuser | admit* |
| 10:26.45 | WIMPy | Unless you explicitly want to set it to something else. |
| 10:27.12 | angryuser | WIMPy, in this case the latest sangoma driver do not generate the file as it should |
| 10:27.49 | WIMPy | The real issue is that chan_dahdi has a bad default. |
| 10:28.10 | WIMPy | If you don't configure it, it should default to unknown, obviousely. |
| 10:28.22 | angryuser | WIMPy, not sure to understand you. about bad default |
| 10:28.56 | angryuser | WIMPy, i will report to sangoma tech's |
| 10:29.02 | WIMPy | Well, you didn't set the value at all, did you? |
| 10:29.16 | WIMPy | So why does it default to anything else than unknown? |
| 10:29.42 | WIMPy | Unknown should work most of the times. |
| 10:29.56 | angryuser | WIMPy, right |
| 10:30.10 | angryuser | WIMPy, by defaut it is set to what ? |
| 10:30.45 | WIMPy | I think national. You could have seen it in your debug :-) |
| 10:31.17 | angryuser | WIMPy, hmmmmmm, yes, national something |
| 10:31.44 | angryuser | WIMPy, was too much focused on 28 cause code |
| 10:32.06 | angryuser | WIMPy, thank you very much |
| 10:32.09 | WIMPy | Well, that was actually quite specific, I think. |
| 10:32.25 | WIMPy | The type of number is part of the number. |
| 10:32.43 | angryuser | WIMPy, just want that you know that you repaired a PBX on Reunion Island in Indian Ocean |
| 10:33.04 | angryuser | Next to madagascart |
| 10:33.08 | angryuser | madagascar |
| 10:33.51 | WIMPy | Hmm. that rings a bell. |
| 10:34.22 | WIMPy | Did we have exchange students from Reunion here? |
| 10:34.41 | angryuser | I am working on remote, i am not there |
| 10:35.06 | angryuser | ITs a beatifull volcanic island |
| 10:35.28 | WIMPy | Google doen't know anything about that. So either it's dumb or I have no idea, what it was :-) |
| 10:36.10 | angryuser | Show madagascar |
| 10:36.23 | angryuser | And it is at right of madagascar |
| 10:36.29 | angryuser | on the middle |
| 10:37.49 | WIMPy | If I win the lottery, I'll take a look :-)) |
| 10:38.36 | irroot | WIMPy lots of south africans holiday there always specials |
| 10:39.56 | angryuser | irroot, WIMPy i wish i go there one day |
| 10:50.18 | *** part/#asterisk pietro (~pietro@88-149-227-2.dynamic.ngi.it) |
| 11:13.00 | cusco | hey folks |
| 11:13.03 | cusco | using ael |
| 11:13.06 | cusco | is this wrong? |
| 11:13.07 | cusco | if("${DIALSTATUS}"=!"ANSWER" && ${EPOCH}<${endDialOut}){ |
| 11:13.22 | cusco | the smaller than |
| 11:13.33 | cusco | im not gettin in the if |
| 11:35.15 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 11:35.15 | *** mode/#asterisk [+o file] by ChanServ |
| 11:40.51 | kaldemar | cusco: is "${DIALSTATUS}"=!"ANSWER" working? |
| 11:43.14 | cusco | kaldemar: i was just looking at it, and it is not |
| 11:43.20 | cusco | I misplaced the != =! |
| 11:43.29 | cusco | rechecking |
| 11:45.56 | cusco | yea,.. thanks |
| 11:46.08 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 11:48.22 | *** join/#asterisk singler (~singler@173.239.155.74) |
| 12:00.17 | jkroon | http://pastebin.com/bwkj5f8Q |
| 12:00.25 | jkroon | can anybody please look at what's wrong there? |
| 12:00.40 | jkroon | qualify=yes on the yealink phone in question does NOT work. |
| 12:01.16 | jkroon | if I need to guess the OK response isn't recognized by asterisk due to the From: line being different? |
| 12:01.20 | jkroon | this is asterisk 1.8.6.0 |
| 12:03.51 | *** join/#asterisk zooz (~zooz@host86-164-219-4.range86-164.btcentralplus.com) |
| 12:03.54 | zooz | hi people |
| 12:04.00 | zooz | does asterisk support SIP over TCP? |
| 12:04.15 | WIMPy | Yes |
| 12:04.28 | WIMPy | At least since 1.8 IIRC. |
| 12:04.33 | jkroon | correct. |
| 12:04.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:04.46 | zooz | checks what version he runs |
| 12:05.16 | zooz | asterisk-1.6.2.20-1 |
| 12:05.17 | zooz | too old |
| 12:05.42 | *** join/#asterisk roxdragon (~gianni@unaffiliated/roxdragon) |
| 12:05.44 | roxdragon | hi all |
| 12:05.49 | roxdragon | exten => *000,3,System(/usr/bin/curl http://192.168.1.30/?L=1) it's possible? |
| 12:08.47 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:08.55 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:09.04 | kaldemar | roxdragon: sure, but there's also func CURL. |
| 12:10.42 | roxdragon | ok |
| 12:10.47 | roxdragon | WARNING[1529]: app_system.c:134 system_exec_helper: Unable to execute '/usr/bin/curl |
| 12:11.37 | jkroon | does someone happen to have firmware for a yealink t10p lying around? |
| 12:13.50 | irroot | jkroon lol nah i need to get the firmwares |
| 12:14.09 | irroot | they not bad phones when you get em working but PITA to get usable |
| 12:15.14 | jkroon | well, this one keels over with qualify=yes |
| 12:17.06 | jkroon | is there any way to request asterisk to use _ANY_ response it receives back in response to OPTIONS request going out as a response? |
| 12:18.26 | kaldemar | roxdragon: do you have /usr/bin/curl in the system? |
| 12:19.27 | roxdragon | yes |
| 12:20.21 | roxdragon | :/usr/bin# curl |
| 12:20.21 | roxdragon | curl: try 'curl --help' or 'curl --manual' for more information |
| 12:20.36 | roxdragon | why? |
| 12:21.04 | kaldemar | that's not proof of having /usr/bin/curl. that's proof of having curl somewhere in path. |
| 12:21.30 | kaldemar | what does CLI say before that warning? |
| 12:22.21 | irroot | roxdragon ./curl in that path or /usr/bin/curl |
| 12:22.43 | irroot | my guess is that its in /usr/local/bin/curl perhaps |
| 12:22.45 | roxdragon | ./curl work |
| 12:23.03 | roxdragon | Dialplan reloaded. |
| 12:23.04 | roxdragon | [Oct 12 14:13:16] WARNING[1562]: app_system.c:134 system_exec_helper: Unable to execute 'curl http://192.168.1.30/?L=1' |
| 12:23.16 | roxdragon | permission? |
| 12:23.19 | kaldemar | that's the only output you see? |
| 12:23.36 | kaldemar | "core set verbose 10" and try again |
| 12:24.05 | roxdragon | -rwxr-xr-x 1 root root 116632 26 giu 19.36 curl |
| 12:24.20 | roxdragon | asterisk:asterisk? |
| 12:24.34 | WIMPy | Looks like it tries to start the whole thing including parameter. |
| 12:24.44 | WIMPy | I think I had that issue before. |
| 12:24.51 | roxdragon | *CLI> core set verbose 10 |
| 12:24.51 | roxdragon | Verbosity was 0 and is now 10 |
| 12:25.12 | roxdragon | now? |
| 12:27.20 | kaldemar | try again |
| 12:27.58 | kaldemar | but i think WIMPy got it. |
| 12:28.33 | roxdragon | restart asterisk? |
| 12:28.43 | kaldemar | no, dial again. |
| 12:29.16 | kaldemar | try System(/usr/bin/curl http:\/\/92.168.1.30\/?L=1) |
| 12:29.19 | roxdragon | ok |
| 12:29.37 | roxdragon | kaldemar, http://codepad.org/mknDuBVj |
| 12:30.03 | kaldemar | looks like System does not like /'s in an argument. |
| 12:30.03 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:31.21 | roxdragon | this? exten => *000,3,System(curl http://192.168.1.30/?L=1) |
| 12:31.21 | *** join/#asterisk atan (atan@unaffiliated/atan) |
| 12:31.39 | WIMPy | quote the parameter? |
| 12:31.44 | jkroon | should chan_sip honestly use all of the call-id, from, to and cseq headers to find a specific call? |
| 12:31.57 | roxdragon | yes |
| 12:32.01 | jkroon | and never mind answering that - yes, it's the right thing to do :( |
| 12:32.06 | jkroon | screw you yealink!! |
| 12:32.41 | jkroon | they change the From: header in response to OPTIONS packets (on the returnnig 200 OK packet), in at least the firmware for the T10p (discontinued phone) |
| 12:34.01 | [TK]D-Fender | roxdragon, specify the full path to curl. |
| 12:35.30 | roxdragon | [TK]D-Fender, exten => *000,3,System(/usr/bin/curl http://192.168.1.30/?L=1) |
| 12:35.34 | roxdragon | don't work |
| 12:36.41 | roxdragon | if I digit curl http://192.168.1.30?L=1 , |
| 12:36.43 | roxdragon | this work |
| 12:36.48 | roxdragon | but on asterisk no |
| 12:41.40 | [TK]D-Fender | roxdragon, Show me |
| 12:41.54 | roxdragon | ok [TK]D-Fender |
| 12:42.04 | WIMPy | What about (curl "http://192.168.1.30?L=1") |
| 12:42.51 | roxdragon | this is arduino... led on or led off L=2 |
| 12:47.59 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
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| 13:17.01 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 13:17.06 | raden | Naikrovek, Morning |
| 13:17.22 | raden | hugs Katty |
| 13:20.47 | *** join/#asterisk serafie (~erin@nat/digium/x-fqfdtsachbvupwyf) |
| 13:20.49 | Naikrovek | raden: morning |
| 13:22.00 | raden | Naikrovek, configs please :D |
| 13:22.12 | Naikrovek | ... |
| 13:22.16 | Naikrovek | oh yeah the phone things. |
| 13:22.25 | Naikrovek | gimme a few minutes to assemble them. |
| 13:22.49 | raden | lol, you were right you would not have remembered lol :) no problem I'm up way early compared to normal |
| 13:24.23 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
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| 13:28.50 | *** part/#asterisk zooz (~zooz@host86-164-219-4.range86-164.btcentralplus.com) |
| 13:30.23 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:30.27 | Naikrovek | raden: this will take a few moments. i'll email you when it's ready. |
| 13:30.58 | raden | no prob bro, dont need it for a good hour |
| 13:44.11 | *** join/#asterisk coppice (~chatzilla@183178203074.ctinets.com) |
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| 13:52.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:57.00 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 13:57.49 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:01.44 | Naikrovek | raden: sent. |
| 14:07.37 | *** join/#asterisk master_of_master (~master_of@p57B53BA7.dip.t-dialin.net) |
| 14:09.23 | raden | Awesome |
| 14:09.36 | raden | my email must be on vacation lol |
| 14:10.34 | Naikrovek | i sent it to your gmail |
| 14:13.29 | *** join/#asterisk jorhaze (~jhazelhof@a80-127-137-124.adsl.xs4all.nl) |
| 14:14.13 | jorhaze | hi, anyone have any idea why asterisk turns my console text to light grey? I do not like light grey. |
| 14:14.55 | Naikrovek | so white text stands out? i dunno. |
| 14:16.07 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 14:17.23 | luckman212 | any Polycom dudes in here |
| 14:17.43 | Naikrovek | Maybe. Ask your question. |
| 14:18.36 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 14:22.16 | jorhaze | I compiled asterisk on debian squeeze and when I run it: /usr/sbin/asterisk - it turns the console text light grey |
| 14:22.37 | jorhaze | the latest, 1.8.7.0 |
| 14:23.16 | jorhaze | on squeeze 6.0.2 |
| 14:24.10 | luckman212 | I just wanted to play with UCS 4.0 that was released about a week ago. But Polycom isn't posting the download publicly, they told me to "get it from my Polycom Reseller" |
| 14:24.23 | luckman212 | was wondering if anyone in here had access to the polycom portal |
| 14:24.43 | Naikrovek | luckman212: ah. there are resellers in here but i don't remember who they are. |
| 14:24.44 | raden | Naikrovek, Still nada |
| 14:24.47 | Naikrovek | they'll contact you. |
| 14:24.49 | Naikrovek | still nada what |
| 14:25.19 | raden | luckman212, call VOIPSUPPLY ask for john |
| 14:25.23 | luckman212 | Naikrovek: yep that's why I was asking.. I figured someone in here might have access to those |
| 14:25.42 | luckman212 | I bought my polycoms from Voiplink but I have no luck reaching them |
| 14:25.59 | raden | luckman212, typical reseller |
| 14:26.11 | raden | I need to start reselling polycom have not had time to get setup with them |
| 14:26.12 | luckman212 | raden: ok will try that, but not sure if they'd want to help me since I didn't buy the phones from there. maybe if I buy one hehe |
| 14:26.26 | raden | Im a aastra resseller but abandoned them after the asterisk 1.8 BS |
| 14:26.48 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 14:26.48 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 14:27.19 | raden | luckman212, buy one and they will help u |
| 14:27.22 | luckman212 | raden: I'm just using them at home with my little asterisk server here, but I really like the polycoms. I have used Aastra too and the poly's just sound better and "feel" better |
| 14:27.50 | raden | luckman212, I like aastra but music on hold and a few things dont work in 1.8 and it only aastra phones |
| 14:28.32 | luckman212 | raden: how could moh not working be a function of the phone? is it a problem with the sip reinvite or something? |
| 14:29.28 | raden | luckman212, something to do with the sip header on aastra phones |
| 14:29.41 | raden | I have had 5-6 people work on it and converted 3 companies back to 1.6 |
| 14:29.50 | raden | switched to polycom |
| 14:29.59 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
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| 14:38.54 | raden | Naikrovek, Wanna resend I got nothing :( brb |
| 14:47.23 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
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| 14:59.54 | r0m|u | p3nguin, you avail? |
| 15:02.34 | *** join/#asterisk jetlag (~jetlag@pool-71-168-246-8.cmdnnj.east.verizon.net) |
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| 15:18.19 | jeffspeff | i'm having a problem when people are calling in to my * system from outside, they randomly get a busy/congested tone. when looking at the cli the call appears to be coming through fine and actually progresses through the dial plan until it hits a Hangup(). I've contacted my sip provider who then identified a "network loop" on their end which they resolved and requested that i restart my asterisk service. i did th |
| 15:18.20 | jeffspeff | is and i'm still having the problem. they continue to maintain that they only see congestion on my end. however i don't see any errors or warnings about a call being congested. how can i check this, and is there anything else that i can look at to resolve this issue? |
| 15:19.37 | [TK]D-Fender | No. * CLI is is. Look at calls. If you can't see it, show us. |
| 15:19.42 | [TK]D-Fender | it* |
| 15:20.40 | raden | jeffspeff, rtp ports forwarded ? |
| 15:20.48 | r1ppa | Can I add a second email address to voicemail addys in voicemail.conf? |
| 15:21.07 | raden | r1ppa, try it and see what happens |
| 15:21.16 | r1ppa | I know a distribution group or a simple alias should do, but curious if voicemail.conf can allow for multiple email addys |
| 15:21.26 | Kobaz | are there settings for which asterisk will start rejecting sip calls |
| 15:21.48 | Kobaz | under medium load i'll sometimes get busy/congestion when the call should have gone through |
| 15:21.50 | jeffspeff | raden, yes they're forwarded. this * box has been in service at this location for almost 2 months now and we are just starting to have this issue |
| 15:22.00 | raden | Kobaz, there is a max call per ext but not set by default to my knowledge should bump to voicemail |
| 15:22.14 | raden | jeffspeff, you running out of channels ? |
| 15:22.15 | *** join/#asterisk [Outcast] (~outcast@westford-nat.juniper.net) |
| 15:22.22 | [TK]D-Fender | r1ppa, There is the primary, and the pager adderss. Anything more you will have to configure your MTA to handle |
| 15:22.28 | Kobaz | raden: well it's not going to go to voicemail unless I send it to voicemail |
| 15:22.35 | Kobaz | raden: but the call never makes it to dialplan |
| 15:22.45 | raden | Kobaz, what does console say |
| 15:22.49 | Kobaz | sip will complain about some max retries |
| 15:23.00 | r1ppa | [TK]D-Fender: ok thanks, aliases it is then! |
| 15:23.01 | Kobaz | i don't have the log item in front of me right now |
| 15:23.03 | raden | Im about ready to make a page where people fill out there system information before we help them LOL |
| 15:23.07 | [TK]D-Fender | <Kobaz> are there settings for which asterisk will start rejecting sip calls |
| 15:23.07 | [TK]D-Fender | <PROTECTED> |
| 15:23.18 | [TK]D-Fender | Kobaz, Answer too slow for the other side's liking perhaps |
| 15:23.27 | Kobaz | two asterisk'es |
| 15:23.36 | jeffspeff | raden, i didn't think that my box would max out on channels itself. i have 50 channels through my provider and only 15 users on my system. i don't believe it's a channels thing. is there a way i can check that? |
| 15:23.42 | Kobaz | yeah i'm thinking the answer winds up being too slow |
| 15:23.49 | Kobaz | any way to lengthen the timeout? |
| 15:23.55 | [TK]D-Fender | Kobaz, and that is * complainign about the other side being too slow (or never getting a response for any other reason) |
| 15:23.55 | Kobaz | without some hacking |
| 15:23.59 | raden | jeffspeff, what kinda router u have ? |
| 15:24.08 | raden | jeffspeff, enough rtp ports forwarded |
| 15:24.34 | [TK]D-Fender | jeffspeff, Go look at the failure |
| 15:24.45 | Kobaz | asterisk is about 10-20% cpu and will sometimes not accept calls |
| 15:24.58 | jeffspeff | [TK]D-Fender, i'm not showing any type of failure on my box though. |
| 15:25.08 | [TK]D-Fender | jeffspeff, Show us the call fail |
| 15:25.19 | [TK]D-Fender | jeffspeff, it dies. You've said so. Show us |
| 15:25.30 | raden | jeffspeff, increase verbose |
| 15:25.40 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 15:25.49 | raden | Kobaz, whats console say when it wont accept calls ? |
| 15:26.20 | Kobaz | [10 12 11:22] <Kobaz> sip will complain about some max retries[10 12 11:23] <Kobaz> i don't have the log item in front of me right now |
| 15:26.22 | jeffspeff | [TK]D-Fender, i'll have to wait for somebody else to report it. and try to grab it. i can't replicate the issue either. there's just enough people complaining about it to make it probable. |
| 15:26.41 | raden | [TK]D-Fender, you ever mess with CIsco voip ? like running off a switch or however they do it ? |
| 15:27.05 | Naikrovek | i have |
| 15:27.05 | [TK]D-Fender | raden, No |
| 15:27.10 | raden | jeffspeff, what is there complaint |
| 15:27.16 | jeffspeff | raden, using sonic wall routers, and we have 10 thousand rtp ports forwarded |
| 15:27.35 | raden | jeffspeff, WTF do you have 10,000 ports forwarded ? |
| 15:28.04 | jeffspeff | raden, the complaint is that when they call our DID # it either instantly hangs up on them or they get a busy signal. they have to try 5 or 6 times to get through. |
| 15:28.15 | raden | Naikrovek, just curious how do them setups work is everything done on a switch or router for call routing ? |
| 15:28.25 | jeffspeff | raden, this is a quote from our sip provider "Please open ports 10K to 20K, UDP for SIP RTP traffic. Thank you. " |
| 15:28.40 | raden | Naikrovek, went and looked at a place they have old cisco stuff and its all just plugged into a network rack and a T1 |
| 15:28.56 | Naikrovek | raden: yep, switches and routers. the configuration is done by software running on a PC. Cisco Call Manager, it's called. |
| 15:29.24 | raden | jeffspeff, if there is a busy signal annd your box is not showing anything on the console its more than likely 95% of the time the SIP provider |
| 15:29.32 | jeffspeff | Naikrovek, CCM is not fun. are you integrating that with asterisk or something? |
| 15:29.35 | Naikrovek | the routers have to be ISR routers (since they require hardware support for voip phones) and there has to be certain switches as well. |
| 15:29.39 | raden | CALL-Centric and Gafachi are notorious for this |
| 15:29.51 | Naikrovek | jeffspeff: no, raden was asking about cisco voip. I've played with it a tiny wee bit but not much. |
| 15:30.08 | raden | Naikrovek, Im going to setup a cisco phone lab in a few months in that case |
| 15:30.28 | Naikrovek | neat. |
| 15:30.37 | jeffspeff | raden, thats what i was thinking, but they say the congestion is on my end. we have more than enough bandwidth, established static routes between our box and their servers, done all we can think of to make it work as best as possible. |
| 15:30.40 | Naikrovek | of course #cisco will offer more help to you than I can. |
| 15:31.31 | raden | jeffspeff, upgrade your router firmware |
| 15:31.50 | jeffspeff | raden, already done, we keep this network updated |
| 15:31.52 | raden | in asterisk re define rtp port range to about 100 ports |
| 15:31.58 | Naikrovek | that's an odd thing to suggest given that we have no actual evidence of the problem yet |
| 15:32.00 | raden | that will support 50 calls |
| 15:32.09 | raden | Naikrovek, have had similar issues |
| 15:33.21 | raden | i hate sonicwalls for VOIP nothing but issues as bad as the netgears i have used that were for VOIP |
| 15:34.30 | raden | jeffspeff, and get some pastes if u need anymore help cause we really just don't know whats wrong as Naikrovek stated |
| 15:35.05 | Naikrovek | yeah. sonicwalls are murder for some reason |
| 15:35.15 | jeffspeff | raden, will try to do. i don't have much info either. just thought i'd ask the audience for any suggestions. |
| 15:36.16 | raden | jeffspeff, you have a old cisco router laying around 2951 or something ? try throwing that in place for a while |
| 15:36.43 | raden | jeffspeff, id say almost 3/4 of the time I have had issues like this has been router issues |
| 15:37.04 | Qwell | I need somebody with an AsteriskNOW install that's willing to do a little test for me. Any takers? |
| 15:37.58 | jeffspeff | raden, lol, not an option. too many routes, vpn's etc. this is not a large network, but good sized. also the sonicwalls do our high availability fail over. tossing a different router in just isn't feasible. |
| 15:38.28 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:38.28 | raden | jeffspeff, DMZ the asterisk box |
| 15:38.47 | raden | or get a dedicated ip and a solid static router |
| 15:38.49 | raden | route |
| 15:38.55 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 15:43.28 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:49.36 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:51.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:53.11 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:04.34 | r0m|u | p3nguin, you avail? |
| 16:06.14 | r0m|u | p3nguin, your sim should have arrived by now. the sime didnt make it out till monday due to issues at the campus post office. but it should have been there by now. please let me know. |
| 16:09.14 | p3nguin | I'll check today. |
| 16:14.16 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:18.36 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 16:18.46 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 16:19.01 | *** join/#asterisk irroot (~irroot@197.105.135.199) |
| 16:23.55 | p3nguin | Hmm, wait... the P.O. was closed on Monday, so it couldn't have gone out until yesterday. |
| 16:24.36 | p3nguin | I'll still check today anyway. |
| 16:26.50 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 16:30.34 | r0m|u | cool. Thanks! |
| 16:32.09 | Gugge | can i use Dial to dial a sip endpoint that requires a password, without adding the endpoint in sip.conf? |
| 16:34.48 | [TK]D-Fender | Gugge, Dial(SIP/user:pass@host/extension) |
| 16:36.04 | Gugge | i thought that worked too, but maybe it doesnt in 10 beta |
| 16:36.26 | Gugge | ill have to try in another version :) |
| 16:36.46 | [TK]D-Fender | It does |
| 16:37.15 | [TK]D-Fender | "think" and "maybe" should be upgraded to "try" and "here's debug from an attempt" |
| 16:37.31 | [TK]D-Fender | You get a lot more with that upgrade.... |
| 16:37.34 | Gugge | sec :) |
| 16:39.55 | Gugge | http://paste2.org/p/1705571 |
| 16:40.21 | Gugge | Dial(SIP/12345678:password@sip2.maxtel.dk/87654321) - but its inviting 12345678 and not 87654321 |
| 16:40.52 | Gugge | and never trying again after unauth |
| 16:41.06 | tuxx- | hey guys. I'm doing an attended transfer of a channel, and that channel comes into the Park() application. But somehow the musiconhold quits on the transfered channel... I'll post a log in a short time. |
| 16:41.17 | p3nguin | I would have thought inviting the user at the host would make sense. |
| 16:41.42 | Gugge | p3nguin: the extension should show up somewhere in the invite pkg though :) |
| 16:42.19 | [TK]D-Fender | Gugge, swap for Dial(SIP/user:pass:exten@host) |
| 16:42.35 | [TK]D-Fender | Gugge, the first should be valid, but I have heard of cases where it didn't functionas it should... |
| 16:42.50 | Gugge | sec |
| 16:43.38 | Gugge | http://paste2.org/p/1705577 |
| 16:43.41 | Gugge | same |
| 16:44.01 | p3nguin | How many devices should the single FXS port on the SPA-3102 handle? |
| 16:45.12 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 16:46.53 | p3nguin | Is it good for the whole house, or do I need something more powerful? |
| 16:46.54 | [TK]D-Fender | p3nguin, 7 REN IIRC |
| 16:47.18 | Qwell | p3nguin: Are they powered phones? |
| 16:47.25 | Qwell | Those are like 0.0000000001 REN |
| 16:47.33 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:48.19 | p3nguin | There's at least one not powered, one cordless with TAD (which is obviously powered), and a couple caller ID boxes. |
| 16:48.20 | Qwell | most phones are going to be like 0.5 REN at most |
| 16:48.24 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 16:48.34 | Gugge | [TK]D-Fender: think i should try a 1.8 release? |
| 16:48.48 | [TK]D-Fender | Gugge, I don't believe this is a bug, its just a syntax question |
| 16:49.07 | Gugge | i would hope so :) |
| 16:51.20 | *** join/#asterisk izz3k (~anton@melody.worknet.ru) |
| 16:52.47 | *** join/#asterisk R-Guy (daemon@mony.mcleodnet.com) |
| 16:53.13 | tuxx- | hey guys. I'm doing an attended transfer of a channel, and that channel comes into the Park() application. But somehow the musiconhold quits on the transfered channel... Log right here: http://pastie.org/private/g15rvguoudotykyu6kesg |
| 16:56.24 | *** part/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 17:04.12 | *** join/#asterisk irroot (~irroot@197.105.135.199) |
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| 17:35.55 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:38.57 | pdtpatrick | Question .. regarding voicemail recordings.. in /var/spool/asterisk/voicemail -- is this where the recordings are kept? For instance, when you call and there's a custom voicemail setup, would that be the 0000.wav file ? |
| 17:41.34 | pdtpatrick | ahh nvm i found it |
| 17:44.25 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 17:44.50 | *** part/#asterisk pietro (~pietro@93-62-151-43.ip23.fastwebnet.it) |
| 17:49.51 | cusco | hi |
| 17:51.52 | cusco | ISP offers a voip service... I am registered with them sucessfully, but when I place a call, I get a Forbidden... |
| 17:51.58 | cusco | here is the sip debug: http://paste.debian.net/136034/ |
| 17:53.07 | cusco | what would be a common cause for this? |
| 17:53.48 | *** join/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com) |
| 17:55.12 | [TK]D-Fender | cusco, Reliably Transmitting (NAT) to 213.13.89.67:5070: <- your provider is not behind NAT. . Also, they use 5070 for SIP? Highly irregular. |
| 17:55.35 | cusco | [TK]D-Fender: they do, its their proxy |
| 17:55.42 | cusco | yes im seeing: Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK569f6f09;rport |
| 17:55.45 | [TK]D-Fender | cusco, Contact: <sip:%2B351302031844@192.168.1.3:5060> <-- you also didn't configure your * to work properly from behind NAT. You are passing them an unreachable private IP to contact you |
| 17:56.00 | [TK]D-Fender | ~sipnat |
| 17:56.00 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 17:56.08 | cusco | I have nat=yes on he peer |
| 17:56.13 | cusco | and externaladdr |
| 17:56.17 | cusco | in general |
| 17:56.37 | [TK]D-Fender | it has been done wrong or it would not be showing what it is. |
| 17:56.49 | [TK]D-Fender | review your work a few more times |
| 17:57.12 | cusco | ok, waht is the difference between externaladdr and externip? |
| 17:58.01 | [TK]D-Fender | first.. there is no such thing as "externaladdr" |
| 17:58.35 | [TK]D-Fender | This is not a valid parameter name |
| 17:58.38 | cusco | you're right, I meant externaddr |
| 17:58.45 | [TK]D-Fender | That also is invalid |
| 17:59.05 | cusco | "; a. "externaddr = hostname[:port]" specifies a static address[:port] to |
| 17:59.10 | [TK]D-Fender | "externhost" or "externip". |
| 17:59.14 | cusco | ok |
| 17:59.48 | [TK]D-Fender | cusco, Also these parameters all need to be in the right place. review the guide, and check your configs |
| 18:01.32 | cusco | ok.. |
| 18:01.59 | cusco | also, I can register a peer from outside nat.. so I guess it can be the peer specifics? |
| 18:02.48 | cusco | wich is: http://paste.debian.net/136040/ |
| 18:04.03 | [TK]D-Fender | registername <- also not valid. peers have no relationships to how you register. |
| 18:04.21 | [TK]D-Fender | cusco, pastebin your sip.conf masking only passwords |
| 18:04.27 | cusco | hmm I wondered about it but I saw no warnig |
| 18:04.47 | cusco | ok will do but lots of ;comments |
| 18:04.51 | WIMPy | [TK]D-Fender: Actually there ist that 'callbackextension' thing. So it can have. |
| 18:05.01 | [TK]D-Fender | "Asterisk sip auth=+351302031844" <- also invalid. |
| 18:05.57 | [TK]D-Fender | WIMPy, no seeting in a peer on your server will affect how you register. that is just the REGISTER statement itself |
| 18:06.28 | WIMPy | [TK]D-Fender: Wrong |
| 18:06.40 | WIMPy | You don;t need a register statement to register. |
| 18:07.47 | cusco | http://paste.debian.net/136042/ |
| 18:08.45 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:09.13 | cusco | the register string seems to be ok, sip show registry show my peer as registered |
| 18:09.48 | cusco | Asterisk sip auth=+351302031844 is invalid ? |
| 18:10.05 | cusco | how do I specify the auth username? |
| 18:15.06 | cusco | somehow it worked before I messed arround I guess |
| 18:15.25 | cusco | and I didn't have externip/externhost set then ! |
| 18:15.31 | cusco | and I could dial out via voip |
| 18:15.43 | cusco | via ISP Voip service I mean |
| 18:18.19 | [TK]D-Fender | WIMPy, How so, and since when? |
| 18:18.41 | [TK]D-Fender | <PROTECTED> |
| 18:19.03 | cusco | ah yes.. ok |
| 18:19.19 | cusco | well why am I sending my internal IP? |
| 18:19.44 | WIMPy | [TK]D-Fender: 'callbackextension' and I don't know since when that exists. But I don't know it for long either. |
| 18:19.47 | [TK]D-Fender | cusco, You have not followed the guide. You failed to specify your localnets. |
| 18:20.14 | *** join/#asterisk jcook_5xdata (~jcook_5xd@173.162.32.1) |
| 18:20.18 | [TK]D-Fender | WIMPy, REGISTER has it's own... |
| 18:20.31 | [TK]D-Fender | WIMPy, thats after the host |
| 18:20.51 | WIMPy | Yes, but the idea is that you don;t need seperate register lines any more, |
| 18:21.11 | [TK]D-Fender | WIMPy, must be 1.8 thing |
| 18:21.20 | WIMPy | Most probably. |
| 18:21.22 | jcook_5xdata | is it possible in asterisk 1.6 to have the called party get a feedback beep when they place the calling party n hold? |
| 18:21.38 | [TK]D-Fender | WIMPy, Which is a hybrid version of something I suggested repeatedly over the past 5+ years :) |
| 18:21.56 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:21.56 | [TK]D-Fender | jcook_5xdata, that is up to your phone |
| 18:22.04 | WIMPy | So you're responsible yourself? :-) |
| 18:22.34 | jcook_5xdata | hhmmm, I am using polycom 650 I will look in sip.cfg thanks |
| 18:22.37 | *** join/#asterisk michael-i (~anonymous@204.11.230.58) |
| 18:23.21 | [TK]D-Fender | jcook_5xdata, there is a stock reminder after a minute or two |
| 18:23.41 | [TK]D-Fender | jcook_5xdata, You control the sound and frequency |
| 18:24.49 | jcook_5xdata | [TK]D-Fender, ? this in asterisk or polycom if it in sip.cfg I must not have configure thanks again |
| 18:27.21 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:29.26 | [TK]D-Fender | jcook_5xdata, the phone. Not Asterisk |
| 18:35.11 | jcook_5xdata | [TK]D-Fender, yup I think I found it <call.hold.localReminder.enabled="1"> |
| 18:36.19 | cusco | [TK]D-Fender: I'm still getting forbidden, I set the localnet and externip |
| 18:36.26 | cusco | http://paste.debian.net/136052/ - still has 192.168.1.3 |
| 18:37.11 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 18:37.33 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 18:38.12 | cusco | my sip.conf is now smaller: http://paste.debian.net/136055/ |
| 18:41.37 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 18:55.02 | cusco | ok solved it... |
| 18:55.14 | cusco | core restart now somehow made a difference instead of sipreload |
| 18:55.29 | [TK]D-Fender | cusco, TECHNICALLY SIP RELOAD SHOULD HAVE SOLVED IT... |
| 18:55.32 | [TK]D-Fender | darn caps... |
| 18:55.59 | [TK]D-Fender | cusco, Ok, so by "solved it" do you mean jsut the IP is right now? Or does that also mean that you are succeeding at the overall call? |
| 18:57.32 | cusco | im succeeding |
| 18:57.44 | cusco | thank you [TK]D-Fender |
| 18:57.51 | [TK]D-Fender | cusco, You're welcome. |
| 18:58.21 | cusco | I have another question now... |
| 18:58.41 | cusco | I registered twinkle from work to home's asterisk.... that [100] in the sip.conf |
| 18:58.52 | cusco | now and I can dial out etc |
| 18:58.58 | cusco | but the dtmf isn't working |
| 18:59.29 | cusco | I have dtmfmode=rfc2833 |
| 18:59.54 | [TK]D-Fender | cusco, if that is what twinkle is using then that should be fine. I presume you are testing this with VoiceMailMain... right? |
| 19:00.17 | cusco | actually no, im calling our workline (provided by asterisk too) |
| 19:00.33 | cusco | wich works from regular phone (pstn) |
| 19:00.41 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:00.49 | *** join/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com) |
| 19:01.13 | cusco | at work, all our peers use rfc2833 |
| 19:01.18 | [TK]D-Fender | cusco, How does "calling your work line" relate to "twinkle"? You have too many things involved in your test to isolate a breakage |
| 19:01.28 | [TK]D-Fender | cusco, Test each leg independently |
| 19:01.45 | mountainm2k | Hi all-- is there a CLI command I can run to see which, or at least how many, channels of a PRI are actually in use? I had some users report fast-busy, and when I checked, I got it too... |
| 19:02.12 | [TK]D-Fender | mountainm2k, "dahdi show channels" should |
| 19:02.22 | mountainm2k | ah, but I'm still on zap |
| 19:02.42 | [TK]D-Fender | then "core show channels concise" will give you numbered channels |
| 19:03.33 | mountainm2k | no such command... I'm *also* on really old ABE, which is like 1.2 I think... |
| 19:03.43 | mountainm2k | which is maybe starting to become more of a problem |
| 19:03.44 | Qwell | umm |
| 19:03.53 | mountainm2k | but the upgrade sounds like a lot of work... |
| 19:04.42 | Qwell | do I dare ask what version of ABE? |
| 19:05.09 | mountainm2k | You dare you dare -- its old tho |
| 19:05.13 | mountainm2k | B.2.2.1 |
| 19:05.30 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 19:05.32 | Qwell | YOU ARE A BAD MAN. |
| 19:05.32 | Qwell | runs to the corner and rolls up into a ball |
| 19:05.40 | mountainm2k | lol |
| 19:06.24 | mountainm2k | I still don't have hardware I can build up a new / test machine, and since its production, I don't really want to just dive into the upgrade some night and hope I get it all working |
| 19:06.39 | mountainm2k | But i'd still like to see how many Zap channels are in use :-P |
| 19:06.47 | Qwell | zap show channels |
| 19:07.02 | mountainm2k | So on that, its saying chan 1 and 2 have extensions listed |
| 19:07.26 | mountainm2k | Hmmm, well, maybe that could be right for this time of day -- lotsa ppl on lunch |
| 19:07.46 | [TK]D-Fender | mountainm2k, "show channels concise |
| 19:07.51 | [TK]D-Fender | no "core" |
| 19:07.59 | cusco | [TK]D-Fender: ok so I tested it only with Read() |
| 19:08.08 | cusco | -- User entered nothing. |
| 19:08.24 | [TK]D-Fender | cusco, then your client isn't using what you think it is |
| 19:09.03 | cusco | twinkle is set to auto |
| 19:09.13 | cusco | let me force it to frc |
| 19:09.15 | cusco | rfc |
| 19:09.34 | cusco | nothing still |
| 19:09.57 | cusco | sip show peer 100 shows: DTMFmode : rfc2833 |
| 19:10.30 | cusco | my twinkle is also behind another nat |
| 19:10.35 | cusco | may this be related? |
| 19:10.49 | cusco | tho sip show peer also shows: Addr->IP : 88.157.128.26:5065 |
| 19:13.00 | mountainm2k | TK -- thanks much, that worked... |
| 19:13.16 | mountainm2k | While I'm here, any way from the CLI to show if a ZAP span is in alarm? |
| 19:13.25 | mountainm2k | or do I need to look at zttool for that |
| 19:13.34 | mountainm2k | thinking about nagios-ing that |
| 19:13.41 | [TK]D-Fender | cusco, shouldn't be |
| 19:14.00 | [TK]D-Fender | mountainm2k, "pri show span 1" |
| 19:14.05 | [TK]D-Fender | mountainm2k, grep-able |
| 19:14.15 | WIMPy | mountainm2k: Today we have 'pri show ...'. Try 'zap show ...' |
| 19:14.36 | cusco | ok |
| 19:14.37 | mountainm2k | tk -- thanks, I can grep on Status, thanks that'll work |
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| 19:14.49 | cusco | I'll lookt at it at home... |
| 19:15.11 | mountainm2k | WIMPy, that might be even easier... I'll look at that too... |
| 19:17.13 | p3nguin | cusco: Depending on what version of Asterisk you have, externaddr is the new externip... just in case no one told you earlier. |
| 19:18.39 | p3nguin | seri, r0m|u: It's still not here. |
| 19:18.46 | p3nguin | Maybe tomorrow. |
| 19:23.43 | *** join/#asterisk nighty- (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 19:24.22 | cusco | p3nguin: ok thanks |
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| 19:34.14 | r0m|u | p3nguin, Like you said monday was no mail running "in campus it was for picking up" but It should be there no longer than tomorrow. |
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| 19:55.50 | nny | anyone one proficient with the vmail.cgi interface know why the "forward" button would not work? (It just clicks). Looked in apache error logs etc so far nothing useful. Maybe perl or other dependency I am missing? |
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| 19:56.28 | nny | wait sorry |
| 19:56.32 | nny | pebkac |
| 19:56.36 | nny | carry on |
| 19:57.06 | WIMPy | I always wonder if ool use the keyboard on their backs. |
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| 20:29.39 | [TK]D-Fender | checkout time, later all |
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| 20:32.05 | Kobaz | so how does connected line work when doing an attended transfer |
| 20:32.34 | Kobaz | A calls B, B calls C, B transfers it's call to C... but C sees the callerid of B, not the original caller A |
| 20:32.35 | WIMPy | That's not a connected line thing. That's a transfer thing. |
| 20:32.39 | Kobaz | would connected line fix that? |
| 20:33.22 | Kobaz | well when you do an attended transfer, asterisk doesn't know it's a transfer |
| 20:33.32 | Kobaz | i would think you could update the callerid after the transfer with connected line |
| 20:33.51 | WIMPy | Yes, that's the problem, as usual. |
| 20:34.02 | Kobaz | cisco supports that apparently |
| 20:34.11 | WIMPy | Connected line is on connect only. |
| 20:34.12 | Kobaz | when doing an attended transfer it keeps the callerid |
| 20:34.42 | WIMPy | Every PBX does it. |
| 20:34.49 | Kobaz | except for asterisk |
| 20:35.11 | WIMPy | And connected line is the wrong direction in your question anyway. |
| 20:35.15 | Kobaz | k |
| 20:35.20 | WIMPy | You're asking about callerid. |
| 20:35.27 | Kobaz | yeah |
| 20:35.34 | Kobaz | doesn't connected line update the callerid? |
| 20:35.49 | WIMPy | Yes, but on the caller side. |
| 20:35.58 | Kobaz | ooo |
| 20:36.37 | WIMPy | So that the number you dialled is replaced by the number that actually answered. |
| 20:37.33 | Kobaz | ah |
| 20:37.34 | Kobaz | okay |
| 20:37.38 | Kobaz | never knew really what that did |
| 20:38.24 | WIMPy | Well, in te PSTN or a PBX a single call can involve a lot of numbers being transferred. |
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| 20:48.51 | p3nguin | [Oct 12 15:48:21] WARNING[31591]: chan_gtalk.c:1606 gtalk_write: Asked to transmit frame type slin, while native formats is 0x4 (ulaw) (read/write = ulaw/ulaw) |
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| 20:49.14 | p3nguin | I get a flood of that warning when dialing out through gtalk. |
| 20:49.40 | p3nguin | Probably more than 50 repeats per second. |
| 20:50.08 | p3nguin | Anyone seen it before and/or fixed it? |
| 20:50.47 | p3nguin | Should I switch to codec slin? |
| 20:51.34 | cusco | talking on gtalk... i was testing it |
| 20:51.52 | cusco | i need a externip too in gtalk.conf rigt? |
| 20:52.07 | p3nguin | The call still seems to proceed as usual, but that warning floods the console until I disconnect. |
| 20:52.11 | cusco | or would tha be externaddr too? |
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| 20:52.32 | p3nguin | externaddr or externhost, depending on if you have a static public IP address or not. |
| 20:53.11 | p3nguin | externip has been changed to externaddr. If you use externip, you should see a notice in the console that externip has been deprecated. |
| 20:53.27 | cusco | in gtqlk.conf too? |
| 20:53.48 | p3nguin | I would think so. |
| 20:53.59 | cusco | gtalk.conf? because with jabber de ug i noticed the same pro lem |
| 20:54.04 | p3nguin | I wouldn't know why it would be deprecated in one channel but not in another. |
| 20:54.09 | cusco | sending my nat'ed ip out |
| 20:54.39 | cusco | the conf sample had externaddr insip.conf and externip in gtalk.conf |
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| 20:55.02 | cusco | ok will look at that |
| 20:55.26 | cusco | also in jabber debug i noticed that google was sending me alaw |
| 20:56.53 | p3nguin | I switched gtalk to slin to stop the flooding of the warning, but now when I call it just rings and rings, never reaching the phone I'm calling. |
| 20:57.50 | cusco | i read somethig about that in voip info |
| 20:58.05 | cusco | don't remember was it was regarding tho |
| 20:58.25 | cusco | but that issue was described near the bottom of the page |
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| 21:03.38 | cusco | can gtalk do dtmf's? |
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| 21:13.08 | p3nguin | If I use slin, it just rings and rings. If I use ulaw, the call makes it to the other phone, but I get that flood. |
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| 21:16.23 | ideaman | Can anyone tell me in 1.6 what directory all the sounds like tt-monkeys are? |
| 21:18.22 | Qwell | Didn't you ask this the other day? You were given 2 correct answers. |
| 21:18.35 | ideaman | I got /var/lib/asterisk/sounds |
| 21:18.39 | Qwell | You need to install extra-sounds |
| 21:18.41 | ideaman | but they aren't there |
| 21:18.44 | ideaman | I did that too |
| 21:18.56 | Qwell | How? |
| 21:19.25 | ideaman | Not sure, because before I install anything, tt-monkeys was already there |
| 21:19.48 | ideaman | I was just looking for the directory so I could find the whole list of sounds that are apparently already installed |
| 21:19.51 | Qwell | You installed something, and you aren't sure how you did it? |
| 21:20.17 | ideaman | No, I installed the extra sound package, but that didn't put any sounds in /var/lib/asterisk/sounds |
| 21:20.30 | Qwell | How did you install it? |
| 21:21.00 | ideaman | apt-get install asterisk-sounds-extra |
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| 21:30.45 | ideaman | any further suggestions? |
| 21:32.20 | Qwell | ask dpkg where it put them |
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| 21:36.49 | p3nguin | Is it not possible to not load res_adsi.so? |
| 21:37.13 | p3nguin | noload => res_adsi.so certainly doesn't stop it from being loaded. |
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| 21:38.05 | Qwell | it's possible that some dep is pulling it in |
| 21:38.05 | Qwell | app_adsiprog and...something else both use it |
| 21:38.22 | p3nguin | I'll see if I can track it down. |
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| 21:45.49 | p3nguin | I'm not seeing anything else that depends on res_adsi. |
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| 21:46.55 | Qwell | voicemail |
| 21:47.20 | Qwell | your best bet would be to just not build it |
| 21:48.39 | p3nguin | I had a feeling I was going to be recompiling today. |
| 21:49.16 | p3nguin | menuselect tree doesn't indicate to me that voicemail needs adsi. How did you determine that? |
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| 21:50.10 | wannknowwhy | evening gents ans ladies |
| 21:51.18 | wannknowwhy | i need some help with a t38 passtrough, i have a sip trunk from a provider that supports t38 |
| 21:51.24 | wannknowwhy | here is my setup |
| 21:53.24 | wannknowwhy | SIP trunk(provider)----ulaw---->(asterisk serverA)------ulaw------>(asterisk serverB)-----ulaw------>(ata)----->Fax amchine |
| 21:54.00 | wannknowwhy | can not get this to work, fax call gets answered on fxs(ata) but just fails after a few seconds |
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| 23:15.09 | wonderworld | hi, i am trying to compile app_konference but the make fails. any ideas? http://pastebin.com/x12ygNY9 |
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| 23:55.37 | seather | Having a problem with incoming faxes, app_fax.c says: Error transmitting fax. result=49 the call dropped preamturely. This is an FXO port on a Sangoma A200 with latest wanpipe, asterisk 1.6 (trixbox) anyone know how I can diagnopse? |
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